fix bug, test

* Allow skipping locks, precompile

* fix tests

* version

.JuliaFormatter.toml

@@ -0,0 +1,9 @@
+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

.appveyor.yml

@@ -1,31 +0,0 @@
-# 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%"

.github/pull_request_template.md

@@ -0,0 +1,22 @@
+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.

.github/workflows/CompatHelper.yml

@@ -0,0 +1,25 @@
+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 }}

.github/workflows/Documentation.yml

@@ -0,0 +1,16 @@
+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 }}

.github/workflows/TagBot.yml

@@ -0,0 +1,15 @@
+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 }}

.github/workflows/Tests.yml

@@ -0,0 +1,41 @@
+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

.gitignore

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

.travis.yml

@@ -1,16 +0,0 @@
-# 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())'

Project.toml

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

README.md

@@ -1,17 +1,23 @@
 PortAudio.jl
 ============
 
-[![Build Status](https://travis-ci.org/JuliaAudio/PortAudio.jl.svg?branch=master)](https://travis-ci.org/JuliaAudio/PortAudio.jl)
-[![Build status](https://ci.appveyor.com/api/projects/status/6x1ha7uvrnel060g/branch/master?svg=true)](https://ci.appveyor.com/project/ssfrr/portaudio-jl/branch/master)
+[![Dev](https://img.shields.io/badge/docs-dev-blue.svg)](https://JuliaAudio.github.io/PortAudio.jl/dev)
+[![Tests](https://github.com/JuliaAudio/PortAudio.jl/actions/workflows/Tests.yml/badge.svg)](https://github.com/JuliaAudio/PortAudio.jl/actions/workflows/Tests.yml)
+[![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.
 
 ## Opening a stream
 
-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.
+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.
+If named keyword arguments `latency` or `samplerate` are unspecified, then PortAudio will use device defaults.
 
 ```julia
-PortAudioStream(inchans=2, outchans=2; eltype=Float32, samplerate=48000Hz, latency=0.1, synced=false)
+PortAudioStream(inchans=2, outchans=2; eltype=Float32, samplerate=48000, latency=0.1)
 ```
 
 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
@@ -25,26 +31,34 @@ You can get a list of your system's devices with the `PortAudio.devices()` funct
 
 ```julia
 julia> PortAudio.devices()
-6-element Array{PortAudio.PortAudioDevice,1}:
- PortAudio.PortAudioDevice("AirPlay","Core Audio",0,2,0)
- PortAudio.PortAudioDevice("Built-in Microph","Core Audio",2,0,1)
- PortAudio.PortAudioDevice("Built-in Output","Core Audio",0,2,2)
- PortAudio.PortAudioDevice("JackRouter","Core Audio",2,2,3)
- PortAudio.PortAudioDevice("After Effects 13.5","Core Audio",0,0,4)
- PortAudio.PortAudioDevice("Built-In Aggregate","Core Audio",2,2,5)
+14-element Vector{PortAudio.PortAudioDevice}:
+ "sof-hda-dsp: - (hw:0,0)" 2→2
+ "sof-hda-dsp: - (hw:0,3)" 0→2
+ "sof-hda-dsp: - (hw:0,4)" 0→2
+ "sof-hda-dsp: - (hw:0,5)" 0→2
+ ⋮
+ "upmix" 8→8
+ "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
 
 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.
 
+## 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
 
 ### Set up an audio pass-through from microphone to speaker
@@ -68,7 +82,7 @@ end
 
 ### Open your built-in microphone and speaker by name
 ```julia
-PortAudioStream("Built-in Microph", "Built-in Output") do stream
+PortAudioStream("default", "default") do stream
     write(stream, stream)
 end
 ```
@@ -76,13 +90,18 @@ end
 ### Record 10 seconds of audio and save to an ogg file
 
 ```julia
-julia> using PortAudio, SampledSignals, LibSndFile
+julia> import LibSndFile # must be in Manifest for FileIO.save to work
 
-julia> stream = PortAudioStream("Built-in Microph", 2, 0)
-PortAudio.PortAudioStream{Float32,SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}}
-  Samplerate: 48000 s⁻¹
-  Buffer Size: 4096 frames
-  2 channel source: "Built-in Microph"
+julia> using PortAudio: PortAudioStream
+
+julia> using SampledSignals: s
+
+julia> using FileIO: save
+
+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)
 480000-frame, 2-channel SampleBuf{Float32, 2, SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}}
@@ -95,13 +114,13 @@ julia> close(stream)
 julia> save(joinpath(homedir(), "Desktop", "myvoice.ogg"), buf)
 ```
 
-## Building the shim library
+### Play an audio signal through the default sound output device
 
-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:
-
-* libportaudio
-* make
-* a C compiler (gcc on linux/macOS, mingw64 on Windows)
-* 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`.
-
-To build the shim, go into the `deps/src` directory and type `make`.
+```julia
+using PortAudio, SampledSignals
+S = 8192 # sampling rate (samples / second)
+x = cos.(2pi*(1:2S)*440/S) # A440 tone for 2 seconds
+PortAudioStream(0, 2; samplerate=S) do stream
+    write(stream, x)
+end
+```

docs/Project.toml

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

docs/make.jl

@@ -0,0 +1,12 @@
+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")

docs/src/index.md

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

docs/src/internals.md

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

examples/audiometer.jl

@@ -1,9 +1,11 @@
 using PortAudio
 
-"""Continuously read from the default audio input and plot an
-ASCII level/peak meter"""
+"""
+Continuously read from the default audio input and plot an
+ASCII level/peak meter
+"""
 function micmeter(metersize)
-    mic = PortAudioStream(1, 0; latency=0.1)
+    mic = PortAudioStream(1, 0; latency = 0.1)
 
     signalmax = zero(eltype(mic))
     println("Press Ctrl-C to quit")
@@ -16,28 +18,32 @@ function micmeter(metersize)
     end
 end
 
-"""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"""
+"""
+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
+"""
 function printmeter(metersize, signal, peak)
     # calculate the positions in terms of characters
     peakpos = clamp(round(Int, peak * metersize), 0, metersize)
-    meterchars = clamp(round(Int, signal * metersize), 0, peakpos-1)
-    blankchars = max(0, peakpos-meterchars-1)
+    meterchars = clamp(round(Int, signal * metersize), 0, peakpos - 1)
+    blankchars = max(0, peakpos - meterchars - 1)
 
     for position in 1:meterchars
-        printstyled(">", color=barcolor(metersize, position))
+        printstyled(">", color = barcolor(metersize, position))
     end
 
-    print(" " ^ blankchars)
-    printstyled("|", color=barcolor(metersize, peakpos))
-    print(" " ^ (metersize - peakpos))
+    print(" "^blankchars)
+    printstyled("|", color = barcolor(metersize, peakpos))
+    print(" "^(metersize - peakpos))
 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
-character, which is red."""
+character, which is red.
+"""
 function barcolor(metersize, position)
-    if position/metersize <= 0.5
+    if position / metersize <= 0.5
         :green
     elseif position == metersize
         :red

examples/lilyplay.jl

@@ -1,127 +1,156 @@
-# Thanks to Jiahao Chen for this great example!
+using Distributed, PortAudio
 
-##
-## NOTE: THIS NEEDS TO BE PORTED OVER TO THE NEW ARCHITECTURE
-##
+# 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
+const DEFAULTDEVICE = -1
 
-using AudioIO
-import AudioIO.play
+function paudio()
+    devs = PortAudio.devices()
+    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
 
-type note{S<:Real, T<:Real}
+play(ostream, sample::Array{Float64, 1}) = write(ostream, sample)
+play(ostr, sample::Array{Int64, 1}) = play(ostr, Float64.(sample))
+
+struct Note{S <: Real, T <: Real}
     pitch::S
     duration::T
     sustained::Bool
 end
 
-function play(A::note, samplingfreq::Real=44100, shape::Function=t->0.6sin(t)+0.2sin(2t)+.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]
+function play(
+    ostream,
+    A::Note,
+    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
-        decay_length = int(length(timesamples) * 0.2)
-        v[end-decay_length:end-1] = v[end-decay_length:end-1] .* linspace(1, 0, decay_length)
+        decay_length = div(length(timesamples), 5)
+        v[(end - decay_length):(end - 1)] =
+            v[(end - decay_length):(end - 1)] .* LinRange(1, 0, decay_length)
     end
-    play(v)
+    play(ostream, v)
     sleep(A.duration)
 end
 
-function parsevoice(melody::String; tempo=132, beatunit=4, lyrics=nothing)
-    play([0]) #Force AudioIO to initialize
-    lyrics_syllables = lyrics==nothing? nothing : split(lyrics)
-
+function parsevoice(melody::String; tempo = 132, beatunit = 4, lyrics = nothing)
+    ostream = paudio() # initialize audio for output
+    lyrics_syllables = lyrics == nothing ? nothing : split(lyrics)
+    lyrics_syllables != nothing && (lyrics_syllables[end] *= "\n")
     note_idx = 1
     oldduration = 4
     for line in split(melody, '\n')
-        percent_idx = findfirst(line, '%') #Trim comment
-        percent_idx == 0 || (line = line[1:percent_idx-1])
+        percent_idx = findfirst('%', line) # Trim comment
+        percent_idx == nothing || (line = line[1:(percent_idx - 1)])
         for token in split(line)
-            pitch, duration, dotted, sustained =parsetoken(token)
-            duration==nothing && (duration = oldduration)
+            pitch, duration, dotted, sustained = parsetoken(token)
+            duration == nothing && (duration = oldduration)
             oldduration = duration
             dotted && (duration *= 1.5)
-            if lyrics_syllables!=nothing && 1<=note_idx<=length(lyrics_syllables) #Print the lyrics, omitting hyphens
-                if lyrics_syllables[note_idx][end]=='-'
-                    print(lyrics_syllables[note_idx][1:end-1])
+            if lyrics_syllables != nothing && 1 <= note_idx <= length(lyrics_syllables)
+                # Print the lyrics, omitting hyphens
+                if lyrics_syllables[note_idx][end] == '-'
+                    print(join(split(lyrics_syllables[note_idx][:], "")[1:(end - 1)]), "")
                 else
                     print(lyrics_syllables[note_idx], ' ')
                 end
             end
-            play(note(pitch, (beatunit/duration)*(60/tempo), sustained))
+            play(ostream, Note(pitch, (beatunit / duration) * (60 / tempo), sustained))
             note_idx += 1
         end
-        println()
     end
 end
 
-function parsetoken(token::String, Atuning::Real=220)
+function parsetoken(token, Atuning::Real = 220)
     state = :findpitch
     pitch = 0.0
     sustain = dotted = false
     lengthbuf = Char[]
     for char in token
         if state == :findpitch
-            scale_idx = findfirst('a':'g', char) + findfirst('A':'G', char)
-            if scale_idx!=0
-                const halfsteps = [12, 14, 3, 5, 7, 8, 10]
-                pitch = Atuning*2^(halfsteps[scale_idx]/12)
+            scale_idx =
+                something(findfirst(char, String(collect('a':'g'))), 0) +
+                something(findfirst(char, String(collect('A':'G'))), 0)
+            if scale_idx != 0
+                halfsteps = [12, 14, 3, 5, 7, 8, 10]
+                pitch = Atuning * 2^(halfsteps[scale_idx] / 12)
                 state = :findlength
-            elseif char=='r'
+            elseif char == 'r'
                 pitch, state = 0, :findlength
             else
                 error("unknown pitch: $char")
             end
         elseif state == :findlength
-            if     char == '#' ; pitch *= 2^(1/12) #sharp
-            elseif char == 'b' ; pitch /= 2^(1/12) #flat
-            elseif char == '\''; pitch *= 2        #higher octave
-            elseif char == ',' ; pitch /= 2        #lower octave
-            elseif char == '.' ; dotted = true     #dotted note
-            elseif char == '~' ; sustain = true    #tied note
+            if char == '#'
+                pitch *= 2^(1 / 12) # sharp
+            elseif char == 'b'
+                pitch /= 2^(1 / 12) # flat
+            elseif char == '\''
+                pitch *= 2          # higher octave
+            elseif char == ','
+                pitch /= 2          # lower octave
+            elseif char == '.'
+                dotted = true       # dotted note
+            elseif char == '~'
+                sustain = true      # tied note
             else
                 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 lengthbuf[end-1:end] == "is"
-                        pitch *= 2^(1/12)
-                        lengthbuf = lengthbuf[1:end-2]
-                    elseif lengthbuf[end-1:end] == "es"
-                        pitch /= 2^(1/12)
-                        lengthbuf = lengthbuf[1:end-2]
+                    if lengthbuf[(end - 1):end] == "is"
+                        pitch *= 2^(1 / 12)
+                        lengthbuf = lengthbuf[1:(end - 2)]
+                    elseif lengthbuf[(end - 1):end] == "es"
+                        pitch /= 2^(1 / 12)
+                        lengthbuf = lengthbuf[1:(end - 2)]
                     end
                 end
             end
         end
     end
     #finalize length
-    lengthstr = convert(String, lengthbuf)
-    duration = isempty(lengthstr) ? nothing : parseint(lengthstr)
+    lengthstr = String(lengthbuf)
+    duration = isempty(lengthstr) ? nothing : tryparse(Int, lengthstr)
     return (pitch, duration, sustain, dotted)
 end
 
-parsevoice("""
+parsevoice(
+    """
 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
 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""",
-lyrics="""
-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!
-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.
-""")
+    lyrics = """
+      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!
+      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.
+      """,
+)
 
 # And now with harmony!
 
-soprano = @async parsevoice("""
+soprano = @spawn parsevoice(
+    """
 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 = @async parsevoice("""
+alto = @spawn parsevoice("""
 a. a. a. a.  a.  a. a. a~ g8 f#.4 a.  a.  a. a.~ a a8 a4~ a2
 """)
-tenor = @async parsevoice("""
+tenor = @spawn parsevoice("""
 d. d. e. f#. f#. e. d. d~ e8 f#.4 f#. a,. d. d.~ d c#8 c#4 c#2
 """)
-bass = @async parsevoice("""
+bass = @spawn parsevoice("""
 d. d. d. d. a,. a,. a,. b,~ c8 d. a., a., a., a., a, a8, a,4 a,2
 """)
 wait(soprano)
@@ -129,19 +158,21 @@ wait(alto)
 wait(tenor)
 wait(bass)
 
-soprano = @async parsevoice("""
+soprano = @spawn parsevoice(
+    """
 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!")
-alto = @async parsevoice("""
+""",
+    lyrics = "Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!",
+)
+alto = @spawn parsevoice("""
 a.4 a. b. c'. c'. b. a. g. f#. f#. g. f#. g.~ g4 f#8 f#~ f#2
 """)
-tenor = @async parsevoice("""
+tenor = @spawn parsevoice("""
 d.4 d. d. d. d. d. d. d. d. d. c#. d. c#.~ c# d8 d d2
 """)
-bass = @async parsevoice("""
+bass = @spawn parsevoice("""
 d.4 d. d. d. a,. a,. a,. a., a., a., a., a., a.,~ a, a,8 d, d,2
 """)
-
 wait(soprano)
 wait(alto)
 wait(tenor)

examples/measure_latency.jl

@@ -4,15 +4,14 @@ 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))
+        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)
+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()
 
@@ -37,7 +36,7 @@ function measure_latency(in_latency = 0.1, out_latency=0.1; is_warmup = false)
     wait(reader)
     wait(writer)
 
-    recorded = collect(reader.result)[:,1]
+    recorded = collect(reader.result)[:, 1]
 
     close(in_stream)
     close(out_stream)

examples/octave-shift.jl

@@ -0,0 +1,89 @@
+#=
+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)

examples/spectrum.jl

@@ -14,7 +14,7 @@ const fs = Float32[float(f) for f in domain(fft(buf)[fmin..fmax])]
 
 while true
     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

examples/tone-buffer.jl

@@ -0,0 +1,21 @@
+#=
+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)

examples/waterfall_heatmap.jl

@@ -8,8 +8,8 @@ The rightmosts column is discarded and the leftmost column is
 left alone.
 """
 function shift1!(buf::AbstractMatrix)
-    for col in size(buf,2):-1:2
-        @. buf[:, col] = buf[:, col-1]
+    for col in size(buf, 2):-1:2
+        @. buf[:, col] = buf[:, col - 1]
     end
 end
 
@@ -20,7 +20,7 @@ function processbuf!(readbuf, win, dispbuf, fftbuf, fftplan)
     readbuf .*= win
     A_mul_B!(fftbuf, fftplan, readbuf)
     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
 
 function processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
@@ -31,17 +31,17 @@ function processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
 end
 
 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
 M = 200 # amount of history to keep
 src = PortAudioStream(1, 2)
 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
 dispbufs = [zeros(Float32, D, M) for i in 1:5, j in 1:5] # STFT bufs
 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
 for _ in 1:M
@@ -53,7 +53,7 @@ heatmaps = map(enumerate(IndexCartesian(), dispbufs)) do ibuf
     buf = ibuf[2]
 
     # 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
 
 center!(scene)

examples/waterfall_lines.jl

@@ -2,7 +2,7 @@ using Makie, GeometryTypes
 using PortAudio
 
 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
 M = 100 # number of lines to draw
 S = 0.5 # motion speed of lines
@@ -10,19 +10,24 @@ src = PortAudioStream(1, 2)
 buf = Array{Float32}(N)
 fftbuf = Array{Complex{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)
 center!(scene)
 
 ls = map(1:M) do _
     yoffset = to_node(to_value(scene[:time]))
     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
-    l = lines(linspace(0,1,D), 0.0f0, zeros(Float32, D),
-        offset=offset, color=(:black, 0.1))
+    l = lines(
+        linspace(0, 1, D),
+        0.0f0,
+        zeros(Float32, D),
+        offset = offset,
+        color = (:black, 0.1),
+    )
     (yoffset, l)
 end
 
@@ -31,7 +36,7 @@ while isopen(scene[:screen])
         isopen(scene[:screen]) || break
         read!(src, 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]
         push!(yoffset, to_value(scene[:time]))
     end

gen/README.md

@@ -0,0 +1 @@
+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.

gen/generator.jl

@@ -0,0 +1,16 @@
+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)

gen/generator.toml

@@ -0,0 +1,9 @@
+[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

runtests.sh

@@ -1,11 +0,0 @@
-#!/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)'

src/libportaudio.jl

@@ -1,158 +1,328 @@
-# Low-level wrappers for Portaudio calls
+module LibPortAudio
 
-# General type aliases
-const PaTime = Cdouble
-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
+using libportaudio_jll
+export libportaudio_jll
 
-const paNoFlag = PaStreamFlags(0x00)
-
-const PA_NO_ERROR = 0
-const PA_INPUT_OVERFLOWED = -10000 + 19
-const PA_OUTPUT_UNDERFLOWED = -10000 + 20
-
-# sample format types
-const paFloat32 = PaSampleFormat(0x01)
-const paInt32   = PaSampleFormat(0x02)
-const paInt24   = PaSampleFormat(0x04)
-const paInt16   = PaSampleFormat(0x08)
-const paInt8    = PaSampleFormat(0x10)
-const paUInt8   = PaSampleFormat(0x20)
-const paNonInterleaved = PaSampleFormat(0x80000000)
-
-const type_to_fmt = Dict{Type, PaSampleFormat}(
-    Float32 => 1,
-    Int32   => 2,
-    # Int24   => 4,
-    Int16   => 8,
-    Int8    => 16,
-    UInt8   => 3
-)
-
-const PaStreamCallbackResult = Cint
-# Callback return values
-const paContinue = PaStreamCallbackResult(0)
-const paComplete = PaStreamCallbackResult(1)
-const paAbort = PaStreamCallbackResult(2)
-
-"""
-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))))
+function Pa_GetVersion()
+    ccall((:Pa_GetVersion, libportaudio), Cint, ())
 end
 
-# because we're calling Pa_ReadStream and PA_WriteStream from separate threads,
-# we put a mutex around libportaudio calls
-const pamutex = ReentrantLock()
+function Pa_GetVersionText()
+    ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
+end
 
-macro locked(ex)
-    quote
-        lock(pamutex) do
-            $(esc(ex))
-        end
-    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
+
+const PaError = Cint
+
+@enum PaErrorCode::Int32 begin
+    paNoError = 0
+    paNotInitialized = -10000
+    paUnanticipatedHostError = -9999
+    paInvalidChannelCount = -9998
+    paInvalidSampleRate = -9997
+    paInvalidDevice = -9996
+    paInvalidFlag = -9995
+    paSampleFormatNotSupported = -9994
+    paBadIODeviceCombination = -9993
+    paInsufficientMemory = -9992
+    paBufferTooBig = -9991
+    paBufferTooSmall = -9990
+    paNullCallback = -9989
+    paBadStreamPtr = -9988
+    paTimedOut = -9987
+    paInternalError = -9986
+    paDeviceUnavailable = -9985
+    paIncompatibleHostApiSpecificStreamInfo = -9984
+    paStreamIsStopped = -9983
+    paStreamIsNotStopped = -9982
+    paInputOverflowed = -9981
+    paOutputUnderflowed = -9980
+    paHostApiNotFound = -9979
+    paInvalidHostApi = -9978
+    paCanNotReadFromACallbackStream = -9977
+    paCanNotWriteToACallbackStream = -9976
+    paCanNotReadFromAnOutputOnlyStream = -9975
+    paCanNotWriteToAnInputOnlyStream = -9974
+    paIncompatibleStreamHostApi = -9973
+    paBadBufferPtr = -9972
+end
+
+function Pa_GetErrorText(errorCode)
+    ccall((:Pa_GetErrorText, libportaudio), Ptr{Cchar}, (PaError,), errorCode)
 end
 
 function Pa_Initialize()
-    err = @locked ccall((:Pa_Initialize, libportaudio), PaError, ())
-    handle_status(err)
+    ccall((:Pa_Initialize, libportaudio), PaError, ())
 end
 
 function Pa_Terminate()
-    err = @locked ccall((:Pa_Terminate, libportaudio), PaError, ())
-    handle_status(err)
+    ccall((:Pa_Terminate, libportaudio), PaError, ())
 end
 
-Pa_GetVersion() = @locked ccall((:Pa_GetVersion, libportaudio), Cint, ())
+const PaDeviceIndex = Cint
 
-function Pa_GetVersionText()
-    versionPtr = @locked ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
-    unsafe_string(versionPtr)
+const PaHostApiIndex = Cint
+
+function Pa_GetHostApiCount()
+    ccall((:Pa_GetHostApiCount, libportaudio), PaHostApiIndex, ())
 end
 
-# Host API Functions
+function Pa_GetDefaultHostApi()
+    ccall((:Pa_GetDefaultHostApi, libportaudio), PaHostApiIndex, ())
+end
 
-# A Host API is the top-level of the PortAudio hierarchy. Each host API has a
-# unique type ID that tells you which native backend it is (JACK, ALSA, ASIO,
-# etc.). On a given system you can identify each backend by its index, which
-# will range between 0 and Pa_GetHostApiCount() - 1. You can enumerate through
-# all the host APIs on the system by iterating through those values.
-
-# PaHostApiTypeId values
-const pa_host_api_names = Dict{PaHostApiTypeId, String}(
-    0 => "In Development", # use while developing support for a new host API
-    1 => "Direct Sound",
-    2 => "MME",
-    3 => "ASIO",
-    4 => "Sound Manager",
-    5 => "Core Audio",
-    7 => "OSS",
-    8 => "ALSA",
-    9 => "AL",
-    10 => "BeOS",
-    11 => "WDMKS",
-    12 => "Jack",
-    13 => "WASAPI",
-    14 => "AudioScience HPI"
-)
+@enum PaHostApiTypeId::UInt32 begin
+    paInDevelopment = 0
+    paDirectSound = 1
+    paMME = 2
+    paASIO = 3
+    paSoundManager = 4
+    paCoreAudio = 5
+    paOSS = 7
+    paALSA = 8
+    paAL = 9
+    paBeOS = 10
+    paWDMKS = 11
+    paJACK = 12
+    paWASAPI = 13
+    paAudioScienceHPI = 14
+end
 
 mutable struct PaHostApiInfo
-    struct_version::Cint
-    api_type::PaHostApiTypeId
+    structVersion::Cint
+    type::PaHostApiTypeId
     name::Ptr{Cchar}
     deviceCount::Cint
     defaultInputDevice::PaDeviceIndex
     defaultOutputDevice::PaDeviceIndex
 end
 
-Pa_GetHostApiInfo(i) = unsafe_load(@locked ccall((:Pa_GetHostApiInfo, libportaudio),
-                                   Ptr{PaHostApiInfo}, (PaHostApiIndex,), i))
-
-# Device Functions
-
-mutable struct PaDeviceInfo
-    struct_version::Cint
-    name::Ptr{Cchar}
-    host_api::PaHostApiIndex
-    max_input_channels::Cint
-    max_output_channels::Cint
-    default_low_input_latency::PaTime
-    default_low_output_latency::PaTime
-    default_high_input_latency::PaTime
-    default_high_output_latency::PaTime
-    default_sample_rate::Cdouble
+function Pa_GetHostApiInfo(hostApi)
+    ccall(
+        (:Pa_GetHostApiInfo, libportaudio),
+        Ptr{PaHostApiInfo},
+        (PaHostApiIndex,),
+        hostApi,
+    )
 end
 
-Pa_GetDeviceCount() = @locked ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())
+function Pa_HostApiTypeIdToHostApiIndex(type)
+    ccall(
+        (:Pa_HostApiTypeIdToHostApiIndex, libportaudio),
+        PaHostApiIndex,
+        (PaHostApiTypeId,),
+        type,
+    )
+end
 
-Pa_GetDeviceInfo(i) = unsafe_load(@locked ccall((:Pa_GetDeviceInfo, libportaudio),
-                                 Ptr{PaDeviceInfo}, (PaDeviceIndex,), i))
+function Pa_HostApiDeviceIndexToDeviceIndex(hostApi, hostApiDeviceIndex)
+    ccall(
+        (:Pa_HostApiDeviceIndexToDeviceIndex, libportaudio),
+        PaDeviceIndex,
+        (PaHostApiIndex, Cint),
+        hostApi,
+        hostApiDeviceIndex,
+    )
+end
 
-Pa_GetDefaultInputDevice() = @locked ccall((:Pa_GetDefaultInputDevice, libportaudio),
-                                   PaDeviceIndex, ())
+mutable struct PaHostErrorInfo
+    hostApiType::PaHostApiTypeId
+    errorCode::Clong
+    errorText::Ptr{Cchar}
+end
 
-Pa_GetDefaultOutputDevice() = @locked ccall((:Pa_GetDefaultOutputDevice, libportaudio),
-                                    PaDeviceIndex, ())
+function Pa_GetLastHostErrorInfo()
+    ccall((:Pa_GetLastHostErrorInfo, libportaudio), Ptr{PaHostErrorInfo}, ())
+end
 
-# Stream Functions
+function Pa_GetDeviceCount()
+    ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())
+end
 
-mutable struct Pa_StreamParameters
+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
+    structVersion::Cint
+    name::Ptr{Cchar}
+    hostApi::PaHostApiIndex
+    maxInputChannels::Cint
+    maxOutputChannels::Cint
+    defaultLowInputLatency::PaTime
+    defaultLowOutputLatency::PaTime
+    defaultHighInputLatency::PaTime
+    defaultHighOutputLatency::PaTime
+    defaultSampleRate::Cdouble
+end
+
+function Pa_GetDeviceInfo(device)
+    ccall((:Pa_GetDeviceInfo, libportaudio), Ptr{PaDeviceInfo}, (PaDeviceIndex,), device)
+end
+
+struct PaStreamParameters
     device::PaDeviceIndex
     channelCount::Cint
     sampleFormat::PaSampleFormat
     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
 
 mutable struct PaStreamInfo
@@ -162,119 +332,108 @@ mutable struct PaStreamInfo
     sampleRate::Cdouble
 end
 
-# function Pa_OpenDefaultStream(inChannels, outChannels,
-#                               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[]
+function Pa_GetStreamInfo(stream)
+    ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo}, (Ptr{PaStream},), stream)
 end
 
-function Pa_StartStream(stream::PaStream)
-    err = @locked ccall((:Pa_StartStream, libportaudio), PaError,
-                (PaStream,), stream)
-    handle_status(err)
+function Pa_GetStreamTime(stream)
+    ccall((:Pa_GetStreamTime, libportaudio), PaTime, (Ptr{PaStream},), stream)
 end
 
-function Pa_StopStream(stream::PaStream)
-    err = @locked ccall((:Pa_StopStream, libportaudio), PaError,
-                (PaStream,), stream)
-    handle_status(err)
+function Pa_GetStreamCpuLoad(stream)
+    ccall((:Pa_GetStreamCpuLoad, libportaudio), Cdouble, (Ptr{PaStream},), stream)
 end
 
-function Pa_CloseStream(stream::PaStream)
-    err = @locked ccall((:Pa_CloseStream, libportaudio), PaError,
-                (PaStream,), stream)
-    handle_status(err)
+function Pa_ReadStream(stream, buffer, frames)
+    ccall(
+        (:Pa_ReadStream, libportaudio),
+        PaError,
+        (Ptr{PaStream}, Ptr{Cvoid}, Culong),
+        stream,
+        buffer,
+        frames,
+    )
 end
 
-function Pa_GetStreamReadAvailable(stream::PaStream)
-    avail = @locked ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong,
-                (PaStream,), stream)
-    avail >= 0 || handle_status(avail)
-    avail
+function Pa_WriteStream(stream, buffer, frames)
+    ccall(
+        (:Pa_WriteStream, libportaudio),
+        PaError,
+        (Ptr{PaStream}, Ptr{Cvoid}, Culong),
+        stream,
+        buffer,
+        frames,
+    )
 end
 
-function Pa_GetStreamWriteAvailable(stream::PaStream)
-    avail = @locked ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong,
-                (PaStream,), stream)
-    avail >= 0 || handle_status(avail)
-    avail
+function Pa_GetStreamReadAvailable(stream)
+    ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong, (Ptr{PaStream},), stream)
 end
 
-function Pa_ReadStream(stream::PaStream, buf::Array, frames::Integer,
-                       show_warnings=true)
-    # without disable_sigint I get a segfault with the error:
-    # "error thrown and no exception handler available."
-    # if the user tries to ctrl-C. Note I've still had some crash problems with
-    # ctrl-C within `pasuspend`, so for now I think either don't use `pasuspend` or
-    # don't use ctrl-C.
-    err = disable_sigint() do
-        @tcall @locked ccall((:Pa_ReadStream, libportaudio), PaError,
-                             (PaStream, Ptr{Cvoid}, Culong),
-                             stream, buf, frames)
-    end
-    handle_status(err, show_warnings)
-    err
+function Pa_GetStreamWriteAvailable(stream)
+    ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong, (Ptr{PaStream},), stream)
 end
 
-function Pa_WriteStream(stream::PaStream, buf::Array, frames::Integer,
-                        show_warnings=true)
-    err = disable_sigint() do
-        @tcall @locked ccall((:Pa_WriteStream, libportaudio), PaError,
-                             (PaStream, Ptr{Cvoid}, Culong),
-                             stream, buf, frames)
-    end
-    handle_status(err, show_warnings)
-    err
+function Pa_GetSampleSize(format)
+    ccall((:Pa_GetSampleSize, libportaudio), PaError, (PaSampleFormat,), format)
 end
 
-# function Pa_GetStreamInfo(stream::PaStream)
-#     infoptr = ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo},
-#             (PaStream, ), stream)
-#
-#     unsafe_load(infoptr)
-# end
-#
-# General utility function to handle the status from the Pa_* functions
-function handle_status(err::PaError, show_warnings::Bool=true)
-    if err == PA_OUTPUT_UNDERFLOWED || err == PA_INPUT_OVERFLOWED
-        if show_warnings
-            msg = @locked ccall((:Pa_GetErrorText, libportaudio),
-                        Ptr{Cchar}, (PaError,), err)
-            @warn("libportaudio: " * unsafe_string(msg))
-        end
-    elseif err != PA_NO_ERROR
-        msg = @locked ccall((:Pa_GetErrorText, libportaudio),
-                    Ptr{Cchar}, (PaError,), err)
-        throw(ErrorException("libportaudio: " * unsafe_string(msg)))
+function Pa_Sleep(msec)
+    ccall((:Pa_Sleep, libportaudio), Cvoid, (Clong,), msec)
+end
+
+const paNoDevice = PaDeviceIndex(-1)
+
+const paUseHostApiSpecificDeviceSpecification = PaDeviceIndex(-2)
+
+const paFloat32 = PaSampleFormat(0x00000001)
+
+const paInt32 = PaSampleFormat(0x00000002)
+
+const paInt24 = PaSampleFormat(0x00000004)
+
+const paInt16 = PaSampleFormat(0x00000008)
+
+const paInt8 = PaSampleFormat(0x00000010)
+
+const paUInt8 = PaSampleFormat(0x00000020)
+
+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 # module

src/precompile.jl

@@ -0,0 +1,29 @@
+# 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))
+
+
+
+
+
+

test/runtests.jl

@@ -1,9 +1,65 @@
 #!/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
 
-using PortAudio
-using Test
-
-@testset "PortAudio Tests" begin
+@testset "Tests without sound" begin
     @testset "Reports version" begin
         io = IOBuffer()
         PortAudio.versioninfo(io)
@@ -13,6 +69,188 @@ using Test
     end
 
     @testset "Can list devices without crashing" begin
-        PortAudio.devices()
+        display(devices())
+        println()
     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

test/runtests_local.jl

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