36 changed files with 854 additions and 2459 deletions
--- a/.JuliaFormatter.toml
+++ b/.JuliaFormatter.toml
@ -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
--- a/.github/pull_request_template.md
+++ b/.github/pull_request_template.md
@ -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.
--- a/.github/workflows/CompatHelper.yml
+++ b/.github/workflows/CompatHelper.yml
@ -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 }}
--- a/.github/workflows/Documentation.yml
+++ b/.github/workflows/Documentation.yml
@ -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 }}
--- a/.github/workflows/TagBot.yml
+++ b/.github/workflows/TagBot.yml
@ -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 }}
--- a/.github/workflows/Tests.yml
+++ b/.github/workflows/Tests.yml
@ -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
--- a/.gitignore
+++ b/.gitignore
@ -1,12 +1,8 @@
 *.swp
+*.so
 *.o
 deps/deps.jl
-deps/build.log
-docs/build
 *.wav
 *.flac
 *.cov
 coverage
-deps/usr/lib/pa_shim.so
-deps/usr/lib/pa_shim.dylib
-deps/usr/lib/pa_shim.dll
--- a/.travis.yml
+++ b/.travis.yml
@ -0,0 +1,15 @@
+# Documentation: http://docs.travis-ci.com/user/languages/julia/
+language: julia
+os:
+  - linux
+  - osx
+sudo: required
+julia:
+  - 0.4
+  - 0.5
+notifications:
+  email: false
+script:
+  # we can't actually run on travis, so just make sure it's installable
+ - if [[ -a .git/shallow ]]; then git fetch --unshallow; fi
+ - julia -e 'Pkg.clone(pwd()); Pkg.build("PortAudio"); using PortAudio'
--- a/6
+++ b/6
@ -19,9 +19,3 @@ AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 THE SOFTWARE.
-
-
-suppressor.jl includes code from the Suppressor.jl package, licensed under the
-MIT "Expat" License:
-
-Copyright (c) 2016: Ismael Venegas Castelló.
--- a/Project.toml
+++ b/Project.toml
@ -1,26 +0,0 @@
-name = "PortAudio"
-uuid = "80ea8bcb-4634-5cb3-8ee8-a132660d1d2d"
-repo = "https://github.com/JuliaAudio/PortAudio.jl.git"
-version = "1.3.0"
-
-[deps]
-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"
-Suppressor = "fd094767-a336-5f1f-9728-57cf17d0bbfb"
-
-[compat]
-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"
-
-[targets]
-test = ["Documenter", "LibSndFile", "Test"]
--- a/README.md
+++ b/README.md
@ -1,23 +1,18 @@
 PortAudio.jl
 ============

-[![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.

+Installable on Linux and OSX: [![Build Status](https://travis-ci.org/JuliaAudio/PortAudio.jl.svg?branch=master)](https://travis-ci.org/JuliaAudio/PortAudio.jl)
+
+Installable on Windows: [![Build status](https://ci.appveyor.com/api/projects/status/6x1ha7uvrnel060g/branch/master?svg=true)](https://ci.appveyor.com/project/ssfrr/portaudio-jl/branch/master)
+
 ## 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.
-If named keyword arguments `latency` or `samplerate` are unspecified, then PortAudio will use device defaults.
+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.

 ```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
@ -31,77 +26,51 @@ You can get a list of your system's devices with the `PortAudio.devices()` funct

 ```julia
 julia> PortAudio.devices()
-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
+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)
 ```

+### 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

 ```julia
 stream = PortAudioStream(2, 2)
-try
-    # cancel with Ctrl-C
-    write(stream, stream)
-finally
-    close(stream)
-end
-```
-
-### Use `do` syntax to auto-close the stream
-```julia
-PortAudioStream(2, 2) do stream
-    write(stream, stream)
-end
+write(stream, stream)
 ```

 ### Open your built-in microphone and speaker by name
 ```julia
-PortAudioStream("default", "default") do stream
-    write(stream, stream)
-end
+stream = PortAudioStream("Built-in Microph", "Built-in Output")
+write(stream, stream)
 ```

 ### Record 10 seconds of audio and save to an ogg file

 ```julia
-julia> import LibSndFile # must be in Manifest for FileIO.save to work
+julia> using PortAudio, SampledSignals, LibSndFile

-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> 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> buf = read(stream, 10s)
 480000-frame, 2-channel SampleBuf{Float32, 2, SIUnits.SIQuantity{Int64,0,0,-1,0,0,0,0,0,0}}
@ -109,18 +78,5 @@ julia> buf = read(stream, 10s)
 ▁▄▂▃▅▃▂▄▃▂▂▁▁▂▂▁▁▄▃▁▁▄▂▁▁▁▄▃▁▁▃▃▁▁▁▁▁▁▁▁▄▄▄▄▄▂▂▂▁▃▃▁▃▄▂▁▁▁▁▃▃▂▁▁▁▁▁▁▃▃▂▂▁▃▃▃▁▁▁▁
 ▁▄▂▃▅▃▂▄▃▂▂▁▁▂▂▁▁▄▃▁▁▄▂▁▁▁▄▃▁▁▃▃▁▁▁▁▁▁▁▁▄▄▄▄▄▂▂▂▁▃▃▁▃▄▂▁▁▁▁▃▃▂▁▁▁▁▁▁▃▃▂▂▁▃▃▃▁▁▁▁

-julia> close(stream)
-
 julia> save(joinpath(homedir(), "Desktop", "myvoice.ogg"), buf)
 ```
-
-### Play an audio signal through the default sound output device
-
-```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
-```
--- a/8
+++ b/8
@ -0,0 +1,8 @@
+julia 0.4
+Compat 0.8.8
+BinDeps
+Devectorize
+SampledSignals 0.3.0
+RingBuffers 0.1.0
+@osx Homebrew
+@windows WinRPM
--- a/appveyor.yml
+++ b/appveyor.yml
@ -0,0 +1,30 @@
+environment:
+  matrix:
+  - JULIAVERSION: "julialang/bin/winnt/x86/0.4/julia-0.4-latest-win32.exe"
+  - JULIAVERSION: "julialang/bin/winnt/x64/0.4/julia-0.4-latest-win64.exe"
+  - JULIAVERSION: "julialang/bin/winnt/x86/0.5/julia-0.5-latest-win32.exe"
+  - JULIAVERSION: "julialang/bin/winnt/x64/0.5/julia-0.5-latest-win64.exe"
+
+notifications:
+  - provider: Email
+    on_build_success: false
+    on_build_failure: false
+    on_build_status_changed: false
+
+install:
+# Download most recent Julia Windows binary
+  - ps: (new-object net.webclient).DownloadFile(
+        $("http://s3.amazonaws.com/"+$env:JULIAVERSION),
+        "C:\projects\julia-binary.exe")
+# Run installer silently, output to C:\projects\julia
+  - C:\projects\julia-binary.exe /S /D=C:\projects\julia
+
+build_script:
+# Need to convert from shallow to complete for Pkg.clone to work
+  - IF EXIST .git\shallow (git fetch --unshallow)
+  - C:\projects\julia\bin\julia -e "versioninfo();
+      Pkg.clone(pwd(), \"PortAudio\"); Pkg.build(\"PortAudio\")"
+
+test_script:
+  # can't actually run the test, so just make sure it's installable
+  - C:\projects\julia\bin\julia -e "using PortAudio"
--- a/deps/build.jl
+++ b/deps/build.jl
@ -0,0 +1,26 @@
+using BinDeps
+using Compat
+
+@BinDeps.setup
+
+ENV["JULIA_ROOT"] = abspath(JULIA_HOME, "../../")
+
+# include alias for WinRPM library
+libportaudio = library_dependency("libportaudio", aliases=["libportaudio-2"])
+
+# TODO: add other providers with correct names
+provides(AptGet, "libportaudio2", libportaudio)
+provides(Pacman, "portaudio", libportaudio)
+
+
+@static if is_apple()
+    using Homebrew
+    provides(Homebrew.HB, "portaudio", libportaudio)
+end
+
+@static if is_windows()
+    using WinRPM
+    provides(WinRPM.RPM, "libportaudio2", libportaudio, os = :Windows)
+end
+
+@BinDeps.install @compat(Dict(:libportaudio => :libportaudio, ))
--- a/docs/Project.toml
+++ b/docs/Project.toml
@ -1,2 +0,0 @@
-[deps]
-Documenter = "e30172f5-a6a5-5a46-863b-614d45cd2de4"
--- a/docs/make.jl
+++ b/docs/make.jl
@ -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")
--- a/docs/src/index.md
+++ b/docs/src/index.md
@ -1,10 +0,0 @@
-# Public interface
-
-```@index
-Pages = ["index.md"]
-```
-
-```@autodocs
-Modules = [PortAudio]
-Private = false
-```
--- a/docs/src/internals.md
+++ b/docs/src/internals.md
@ -1,10 +0,0 @@
-# Internals
-
-```@index
-Pages = ["internals.md"]
-```
-
-```@autodocs
-Modules = [PortAudio]
-Public = false
-```
--- a/examples/audiometer.jl
+++ b/examples/audiometer.jl
@ -1,49 +1,43 @@
 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; blocksize=512)

    signalmax = zero(eltype(mic))
    println("Press Ctrl-C to quit")
    while true
        block = read(mic, 512)
-        blockmax = maximum(abs.(block)) # find the maximum value in the block
+        blockmax = maximum(abs(block)) # find the maximum value in the block
        signalmax = max(signalmax, blockmax) # keep the maximum value ever
        print("\r") # reset the cursor to the beginning of the line
        printmeter(metersize, blockmax, signalmax)
    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))
+        print_with_color(barcolor(metersize, position), ">")
    end

-    print(" "^blankchars)
-    printstyled("|", color = barcolor(metersize, peakpos))
-    print(" "^(metersize - peakpos))
+    print(" " ^ blankchars)
+    print_with_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
--- a/examples/lilyplay.jl
+++ b/examples/lilyplay.jl
@ -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
-const DEFAULTDEVICE = -1
+##
+## NOTE: THIS NEEDS TO BE PORTED OVER TO THE NEW ARCHITECTURE
+##

-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
+using AudioIO
+import AudioIO.play

-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}
+type note{S<:Real, T<:Real}
    pitch::S
    duration::T
    sustained::Bool
 end

-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]
+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]
    if !A.sustained
-        decay_length = div(length(timesamples), 5)
-        v[(end - decay_length):(end - 1)] =
-            v[(end - decay_length):(end - 1)] .* LinRange(1, 0, decay_length)
+        decay_length = int(length(timesamples) * 0.2)
+        v[end-decay_length:end-1] = v[end-decay_length:end-1] .* linspace(1, 0, decay_length)
    end
-    play(ostream, v)
+    play(v)
    sleep(A.duration)
 end

-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")
+function parsevoice(melody::String; tempo=132, beatunit=4, lyrics=nothing)
+    play([0]) #Force AudioIO to initialize
+    lyrics_syllables = lyrics==nothing? nothing : split(lyrics)
+
    note_idx = 1
    oldduration = 4
    for line in split(melody, '\n')
-        percent_idx = findfirst('%', line) # Trim comment
-        percent_idx == nothing || (line = line[1:(percent_idx - 1)])
+        percent_idx = findfirst(line, '%') #Trim comment
+        percent_idx == 0 || (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(join(split(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(lyrics_syllables[note_idx][1:end-1])
                else
                    print(lyrics_syllables[note_idx], ' ')
                end
            end
-            play(ostream, Note(pitch, (beatunit / duration) * (60 / tempo), sustained))
+            play(note(pitch, (beatunit/duration)*(60/tempo), sustained))
            note_idx += 1
        end
+        println()
    end
 end

-function parsetoken(token, Atuning::Real = 220)
+function parsetoken(token::String, Atuning::Real=220)
    state = :findpitch
    pitch = 0.0
    sustain = dotted = false
    lengthbuf = Char[]
    for char in token
        if state == :findpitch
-            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)
+            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)
                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 = String(lengthbuf)
-    duration = isempty(lengthstr) ? nothing : tryparse(Int, lengthstr)
+    lengthstr = convert(String, lengthbuf)
+    duration = isempty(lengthstr) ? nothing : parseint(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 = @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
-""",
-    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
 """)
-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
 """)
-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
 """)
 wait(soprano)
@ -158,21 +129,19 @@ wait(alto)
 wait(tenor)
 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
-""",
-    lyrics = "Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!",
-)
-alto = @spawn parsevoice("""
+""", lyrics="Wir be- tre- ten feu- er- trun- ken, Himm- li- sche, dein Hei- - lig- thum!")
+alto = @async parsevoice("""
 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
 """)
-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
 """)
+
 wait(soprano)
 wait(alto)
 wait(tenor)
--- a/examples/measure_latency.jl
+++ b/examples/measure_latency.jl
@ -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
--- a/examples/octave-shift.jl
+++ b/examples/octave-shift.jl
@ -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)
--- a/examples/spectrum.jl
+++ b/examples/spectrum.jl
@ -3,10 +3,10 @@

 module SpectrumExample

-using GR, PortAudio, SampledSignals, FFTW
+using GR, PortAudio, SampledSignals

 const N = 1024
-const stream = PortAudioStream(1, 0)
+const stream = PortAudioStream(1, 0, blocksize=N)
 const buf = read(stream, N)
 const fmin = 0Hz
 const fmax = 10000Hz
@ -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
--- a/examples/tone-buffer.jl
+++ b/examples/tone-buffer.jl
@ -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)
--- a/examples/waterfall_heatmap.jl
+++ b/examples/waterfall_heatmap.jl
@ -1,67 +0,0 @@
-using Makie
-using PortAudio
-using DSP
-
-"""
-Slide the values in the given matrix to the right by 1.
-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]
-    end
-end
-
-"""
-takes a block of audio, FFT it, and write it to the beginning of the buffer
-"""
-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))
-end
-
-function processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
-    read!(src, buf)
-    for dispbuf in dispbufs
-        processbuf!(buf, win, dispbuf, fftbuf, fftplan)
-    end
-end
-
-N = 1024 # size of audio read
-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)
-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))
-
-#pre-fill the display buffer so we can do a reasonable colormap
-for _ in 1:M
-    processblock!(src, buf, win, dispbufs, fftbuf, fftplan)
-end
-
-heatmaps = map(enumerate(IndexCartesian(), dispbufs)) do ibuf
-    i = ibuf[1]
-    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)))
-end
-
-center!(scene)
-
-while isopen(scene[:screen])
-    processblock!(src, buf, dispbufs, fftbuf, fftplan)
-    for (hm, db) in zip(heatmaps, dispbufs)
-        hm[:heatmap] = db
-    end
-    render_frame(scene)
-end
--- a/examples/waterfall_lines.jl
+++ b/examples/waterfall_lines.jl
@ -1,43 +0,0 @@
-using Makie, GeometryTypes
-using PortAudio
-
-N = 1024 # size of audio read
-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
-src = PortAudioStream(1, 2)
-buf = Array{Float32}(N)
-fftbuf = Array{Complex{Float32}}(N2)
-magbuf = Array{Float32}(N2)
-fftplan = plan_rfft(buf; flags = FFTW.EXHAUSTIVE)
-
-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)
-    end
-    l = lines(
-        linspace(0, 1, D),
-        0.0f0,
-        zeros(Float32, D),
-        offset = offset,
-        color = (:black, 0.1),
-    )
-    (yoffset, l)
-end
-
-while isopen(scene[:screen])
-    for (yoffset, line) in ls
-        isopen(scene[:screen]) || break
-        read!(src, buf)
-        A_mul_B!(fftbuf, fftplan, buf)
-        @. magbuf = log(clamp(abs(fftbuf), 0.0001, Inf)) / 10 + 0.5
-        line[:z] = magbuf[1:D]
-        push!(yoffset, to_value(scene[:time]))
-    end
-end
--- a/gen/README.md
+++ b/gen/README.md
@ -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.
--- a/gen/generator.jl
+++ b/gen/generator.jl
@ -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)
--- a/gen/generator.toml
+++ b/gen/generator.toml
@ -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
--- a/runtests.sh
+++ b/runtests.sh
@ -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)'
--- a/src/PortAudio.jl
+++ b/src/PortAudio.jl
--- a/src/libportaudio.jl
+++ b/src/libportaudio.jl
@ -1,439 +1,250 @@
-module LibPortAudio
+# Low-level wrappers for Portaudio calls

-using libportaudio_jll
-export libportaudio_jll
+# General type aliases
+typealias PaTime Cdouble
+typealias PaError Cint
+typealias PaSampleFormat Culong
+typealias PaDeviceIndex Cint
+typealias PaHostApiIndex Cint
+typealias PaHostApiTypeId Cint
+# PaStream is always used as an opaque type, so we're always dealing
+# with the pointer
+typealias PaStream Ptr{Void}
+typealias PaStreamCallback Void
+typealias PaStreamFlags Culong

-function Pa_GetVersion()
-    ccall((:Pa_GetVersion, libportaudio), Cint, ())
-end
+const paNoFlag = PaStreamFlags(0x00)

-function Pa_GetVersionText()
-    ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
-end
+const PA_NO_ERROR = 0
+const PA_INPUT_OVERFLOWED = -10000 + 19
+const PA_OUTPUT_UNDERFLOWED = -10000 + 20

-mutable struct PaVersionInfo
-    versionMajor::Cint
-    versionMinor::Cint
-    versionSubMinor::Cint
-    versionControlRevision::Ptr{Cchar}
-    versionText::Ptr{Cchar}
-end
+# 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)

-# 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 type_to_fmt = Dict{Type, PaSampleFormat}(
+    Float32 => 1,
+    Int32   => 2,
+    # Int24   => 4,
+    Int16   => 8,
+    Int8    => 16,
+    UInt8   => 3
+)

-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
+typealias PaStreamCallbackResult Cint
+# Callback return values
+const paContinue = PaStreamCallbackResult(0)
+const paComplete = PaStreamCallbackResult(1)
+const paAbort = PaStreamCallbackResult(2)

 function Pa_Initialize()
-    ccall((:Pa_Initialize, libportaudio), PaError, ())
+    err = ccall((:Pa_Initialize, libportaudio), PaError, ())
+    handle_status(err)
 end

 function Pa_Terminate()
-    ccall((:Pa_Terminate, libportaudio), PaError, ())
+    err = ccall((:Pa_Terminate, libportaudio), PaError, ())
+    handle_status(err)
 end

-const PaDeviceIndex = Cint
+Pa_GetVersion() = ccall((:Pa_GetVersion, libportaudio), Cint, ())

-const PaHostApiIndex = Cint
-
-function Pa_GetHostApiCount()
-    ccall((:Pa_GetHostApiCount, libportaudio), PaHostApiIndex, ())
+function Pa_GetVersionText()
+    versionPtr = ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
+    unsafe_string(versionPtr)
 end

-function Pa_GetDefaultHostApi()
-    ccall((:Pa_GetDefaultHostApi, libportaudio), PaHostApiIndex, ())
-end
+# Host API Functions

-@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
+# 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.

-mutable struct PaHostApiInfo
-    structVersion::Cint
-    type::PaHostApiTypeId
+# PaHostApiTypeId values
+const pa_host_api_names = Dict{PaHostApiTypeId, UTF8String}(
+    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"
+)
+
+type PaHostApiInfo
+    struct_version::Cint
+    api_type::PaHostApiTypeId
    name::Ptr{Cchar}
    deviceCount::Cint
    defaultInputDevice::PaDeviceIndex
    defaultOutputDevice::PaDeviceIndex
 end

-function Pa_GetHostApiInfo(hostApi)
-    ccall(
-        (:Pa_GetHostApiInfo, libportaudio),
-        Ptr{PaHostApiInfo},
-        (PaHostApiIndex,),
-        hostApi,
-    )
-end
+Pa_GetHostApiInfo(i) = unsafe_load(ccall((:Pa_GetHostApiInfo, libportaudio),
+                                   Ptr{PaHostApiInfo}, (PaHostApiIndex,), i))

-function Pa_HostApiTypeIdToHostApiIndex(type)
-    ccall(
-        (:Pa_HostApiTypeIdToHostApiIndex, libportaudio),
-        PaHostApiIndex,
-        (PaHostApiTypeId,),
-        type,
-    )
-end
+# Device Functions

-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
-    structVersion::Cint
+type PaDeviceInfo
+    struct_version::Cint
    name::Ptr{Cchar}
-    hostApi::PaHostApiIndex
-    maxInputChannels::Cint
-    maxOutputChannels::Cint
-    defaultLowInputLatency::PaTime
-    defaultLowOutputLatency::PaTime
-    defaultHighInputLatency::PaTime
-    defaultHighOutputLatency::PaTime
-    defaultSampleRate::Cdouble
+    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
 end

-function Pa_GetDeviceInfo(device)
-    ccall((:Pa_GetDeviceInfo, libportaudio), Ptr{PaDeviceInfo}, (PaDeviceIndex,), device)
-end
+Pa_GetDeviceCount() = ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())

-struct PaStreamParameters
+Pa_GetDeviceInfo(i) = unsafe_load(ccall((:Pa_GetDeviceInfo, libportaudio),
+                                 Ptr{PaDeviceInfo}, (PaDeviceIndex,), i))
+
+Pa_GetDefaultInputDevice() = ccall((:Pa_GetDefaultInputDevice, libportaudio),
+                                   PaDeviceIndex, ())
+
+Pa_GetDefaultOutputDevice() = ccall((:Pa_GetDefaultOutputDevice, libportaudio),
+                                    PaDeviceIndex, ())
+
+# Stream Functions
+
+type Pa_StreamParameters
    device::PaDeviceIndex
    channelCount::Cint
    sampleFormat::PaSampleFormat
    suggestedLatency::PaTime
-    hostApiSpecificStreamInfo::Ptr{Cvoid}
+    hostAPISpecificStreamInfo::Ptr{Void}
 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
+type PaStreamInfo
    structVersion::Cint
    inputLatency::PaTime
    outputLatency::PaTime
    sampleRate::Cdouble
 end

-function Pa_GetStreamInfo(stream)
-    ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo}, (Ptr{PaStream},), stream)
+# 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{Void}, Ref{Void}),
+#                 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 = ccall((:Pa_OpenStream, libportaudio), PaError,
+                (Ref{PaStream},
+                Ptr{Pa_StreamParameters},
+                Ptr{Pa_StreamParameters},
+                Cdouble, Culong, PaStreamFlags,
+                Ptr{Void}, Ptr{Void}),
+                streamPtr,
+                inParams, outParams,
+                sampleRate, framesPerBuffer, flags,
+                callback, userdata)
+    handle_status(err)
+    streamPtr[]
 end

-function Pa_GetStreamTime(stream)
-    ccall((:Pa_GetStreamTime, libportaudio), PaTime, (Ptr{PaStream},), stream)
+function Pa_StartStream(stream::PaStream)
+    err = ccall((:Pa_StartStream, libportaudio), PaError,
+                (PaStream,), stream)
+    handle_status(err)
 end

-function Pa_GetStreamCpuLoad(stream)
-    ccall((:Pa_GetStreamCpuLoad, libportaudio), Cdouble, (Ptr{PaStream},), stream)
+function Pa_StopStream(stream::PaStream)
+    err = ccall((:Pa_StopStream, libportaudio), PaError,
+                (PaStream,), stream)
+    handle_status(err)
 end

-function Pa_ReadStream(stream, buffer, frames)
-    ccall(
-        (:Pa_ReadStream, libportaudio),
-        PaError,
-        (Ptr{PaStream}, Ptr{Cvoid}, Culong),
-        stream,
-        buffer,
-        frames,
-    )
+function Pa_CloseStream(stream::PaStream)
+    err = ccall((:Pa_CloseStream, libportaudio), PaError,
+                (PaStream,), stream)
+    handle_status(err)
 end

-function Pa_WriteStream(stream, buffer, frames)
-    ccall(
-        (:Pa_WriteStream, libportaudio),
-        PaError,
-        (Ptr{PaStream}, Ptr{Cvoid}, Culong),
-        stream,
-        buffer,
-        frames,
-    )
+function Pa_GetStreamReadAvailable(stream::PaStream)
+    avail = ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong,
+                (PaStream,), stream)
+    avail >= 0 || handle_status(avail)
+    avail
 end

-function Pa_GetStreamReadAvailable(stream)
-    ccall((:Pa_GetStreamReadAvailable, libportaudio), Clong, (Ptr{PaStream},), stream)
+function Pa_GetStreamWriteAvailable(stream::PaStream)
+    avail = ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong,
+                (PaStream,), stream)
+    avail >= 0 || handle_status(avail)
+    avail
 end

-function Pa_GetStreamWriteAvailable(stream)
-    ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong, (Ptr{PaStream},), stream)
+function Pa_ReadStream(stream::PaStream, buf::Array, frames::Integer=length(buf),
+                       show_warnings::Bool=true)
+    frames <= length(buf) || error("Need a buffer at least $frames long")
+    err = ccall((:Pa_ReadStream, libportaudio), PaError,
+                (PaStream, Ptr{Void}, Culong),
+                stream, buf, frames)
+    handle_status(err, show_warnings)
+    buf
 end

-function Pa_GetSampleSize(format)
-    ccall((:Pa_GetSampleSize, libportaudio), PaError, (PaSampleFormat,), format)
+function Pa_WriteStream(stream::PaStream, buf::Array, frames::Integer=length(buf),
+                        show_warnings::Bool=true)
+    frames <= length(buf) || error("Need a buffer at least $frames long")
+    err = ccall((:Pa_WriteStream, libportaudio), PaError,
+                (PaStream, Ptr{Void}, Culong),
+                stream, buf, frames)
+    handle_status(err, show_warnings)
+    nothing
 end

-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
+# 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 = ccall((:Pa_GetErrorText, libportaudio),
+                        Ptr{Cchar}, (PaError,), err)
+            warn("libportaudio: " * bytestring(msg))
+        end
+    elseif err != PA_NO_ERROR
+        msg = ccall((:Pa_GetErrorText, libportaudio),
+                    Ptr{Cchar}, (PaError,), err)
+        error("libportaudio: " * bytestring(msg))
    end
 end
-
-end # module
--- a/src/precompile.jl
+++ b/src/precompile.jl
@ -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))
-
-
-
-
-
-
--- a/test/REQUIRE
+++ b/test/REQUIRE
@ -0,0 +1 @@
+BaseTestNext
--- a/test/runtests.jl
+++ b/test/runtests.jl
@ -1,256 +1,181 @@
 #!/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
+if VERSION >= v"0.5.0-dev+7720"
+    using Base.Test
+else
+    using BaseTestNext
+end
+using PortAudio
+using SampledSignals
+using RingBuffers
+using SampledSignals
+
+function test_callback(inchans, outchans)
+    nframes = Culong(8)
+
+    cb = PortAudio.pa_callbacks[Float32]
+    inbuf = rand(Float32, inchans*nframes) # simulate microphone input
+    sourcebuf = LockFreeRingBuffer(Float32, inchans*nframes*8) # the microphone input should end up here
+
+    outbuf = zeros(Float32, outchans*nframes) # this is where the output should go
+    sinkbuf = LockFreeRingBuffer(Float32, outchans*nframes*8) # the callback should copy this to outbuf
+
+    # 2 input channels, 3 output channels
+    info = PortAudio.CallbackInfo(inchans, sourcebuf, outchans, sinkbuf, true)
+
+    # handle any conversions here so they don't mess with the allocation
+    # the seemingly-redundant type specifiers avoid some allocation during the ccall.
+    # might be due to https://github.com/JuliaLang/julia/issues/15276
+    inptr::Ptr{Float32} = Ptr{Float32}(pointer(inbuf))
+    outptr::Ptr{Float32} = Ptr{Float32}(pointer(outbuf))
+    flags = Culong(0)
+    infoptr::Ptr{PortAudio.CallbackInfo{Float32}} = Ptr{PortAudio.CallbackInfo{Float32}}(pointer_from_objref(info))
+
+    testin = zeros(Float32, inchans*nframes)
+    testout = rand(Float32, outchans*nframes)
+    write(sinkbuf, testout) # fill the output ringbuffer
+    ret = ccall(cb, Cint,
+        (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{PortAudio.CallbackInfo{Float32}}),
+        inptr, outptr, nframes, C_NULL, flags, infoptr)
+    @test ret === PortAudio.paContinue
+    @test outbuf == testout
+    read!(sourcebuf, testin)
+    @test inbuf == testin
+
+    if outchans > 0
+        underfill = 3 # should be less than nframes
+        testout = rand(Float32, outchans*underfill)
+        write(sinkbuf, testout) # underfill the output ringbuffer
+        # call again (partial underrun)
+        ret = ccall(cb, Cint,
+            (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{PortAudio.CallbackInfo{Float32}}),
+            inptr, outptr, nframes, C_NULL, flags, infoptr)
+        @test ret === PortAudio.paContinue
+        @test outbuf[1:outchans*underfill] == testout
+        @test outbuf[outchans*underfill+1:outchans*nframes] == zeros(Float32, (nframes-underfill)*outchans)
+        @test nreadable(sourcebuf) == inchans*underfill
+        @test read!(sourcebuf, testin) == inchans*underfill
+        @test testin[1:inchans*underfill] == inbuf[1:inchans*underfill]
+
+        # call again (total underrun)
+        ret = ccall(cb, Cint,
+            (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{PortAudio.CallbackInfo{Float32}}),
+            inptr, outptr, nframes, C_NULL, flags, infoptr)
+        @test ret === PortAudio.paContinue
+        @test outbuf == zeros(Float32, outchans*nframes)
+        @test nreadable(sourcebuf) == 0
+
+        write(sinkbuf, testout) # fill the output ringbuffer
+        # test allocation
+        alloc = @allocated ccall(cb, Cint,
+            (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{PortAudio.CallbackInfo{Float32}}),
+            inptr, outptr, nframes, C_NULL, flags, infoptr)
+        @test alloc == 0
+        # now test allocation in underrun state
+        alloc = @allocated ccall(cb, Cint,
+            (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{PortAudio.CallbackInfo{Float32}}),
+            inptr, outptr, nframes, C_NULL, flags, infoptr)
+        @test alloc == 0
+    end
+end
+
+# these test are currently set up to run on OSX
+
+@testset "PortAudio Tests" begin
+    devs = PortAudio.devices()
+    i = findfirst(d -> d.maxinchans > 0, devs)
+    indev = i > 0 ? devs[i] : nothing
+    i = findfirst(d -> d.maxoutchans > 0, devs)
+    outdev = i > 0 ? devs[i] : nothing
+    i = findfirst(d -> d.maxoutchans > 0 && d.maxinchans > 0, devs)
+    duplexdev = i > 0 ? devs[i] : nothing
+
    @testset "Reports version" begin
        io = IOBuffer()
        PortAudio.versioninfo(io)
-        result = split(String(take!((io))), "\n")
+        result = takebuf_string(io)
        # make sure this is the same version I tested with
-        @test startswith(result[1], "PortAudio V19")
+        @test result ==
+            """PortAudio V19-devel (built Aug  6 2014 17:54:39)
+            Version Number: 1899
+            """
    end

-    @testset "Can list devices without crashing" begin
-        display(devices())
-        println()
+    @testset "PortAudio Callback works for duplex stream" begin
+        test_callback(2, 3)
    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
+    @testset "Callback works with input-only stream" begin
+        test_callback(2, 0)
    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
+    @testset "Callback works with output-only stream" begin
+        test_callback(0, 2)
    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)
+    @testset "Open Default Device" begin
+        println("Recording...")
+        stream = PortAudioStream(2, 0)
+        buf = read(stream, 5s)
+        close(stream)
+        @test size(buf) == (round(Int, 5 * samplerate(stream)), nchannels(stream.source))
+        println("Playing back recording...")
+        stream = PortAudioStream(0, 2)
+        write(stream, buf)
+        println("flushing...")
+        flush(stream)
+        close(stream)
+        println("Testing pass-through")
+        stream = PortAudioStream(2, 2)
+        write(stream, stream, 5s)
+        flush(stream)
+        close(stream)
+        println("done")
+    end
+    @testset "Samplerate-converting writing" begin
+        stream = PortAudioStream()
+        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("Built-in Microph", "Built-in Output")
+        buf = read(stream, 0.001s)
+        @test size(buf) == (round(Int, 0.001 * samplerate(stream)), nchannels(stream.source))
+        write(stream, buf)
+        io = IOBuffer()
+        show(io, stream)
+        @test takebuf_string(io) == """
+        PortAudio.PortAudioStream{Float32}
+          Samplerate: 48000.0Hz
+          Buffer Size: 4096 frames
+          2 channel sink: "Built-in Output"
+          2 channel source: "Built-in Microph\""""
+        close(stream)
+    end
+    @testset "Error on wrong name" begin
+        @test_throws ErrorException PortAudioStream("foobarbaz")
+    end
+    # 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
+        stream = PortAudioStream(0, 2)
+        buf = SampleBuf(rand(eltype(stream), 48000, nchannels(stream.sink))*0.1, samplerate(stream))
+        t1 = @async write(stream, buf)
+        t2 = @async write(stream, buf)
+        @test wait(t1) == 48000
+        @test wait(t2) == 48000
+        flush(stream)
+        close(stream)
+    end
+    @testset "Queued Reading" begin
+        stream = PortAudioStream(2, 0)
+        buf = SampleBuf(rand(eltype(stream), 48000, nchannels(stream.source))*0.1, samplerate(stream))
+        t1 = @async read!(stream, buf)
+        t2 = @async read!(stream, buf)
+        @test wait(t1) == 48000
+        @test wait(t2) == 48000
+        close(stream)
+    end
 end
--- a/test/runtests_local.jl
+++ b/test/runtests_local.jl
@ -1,95 +0,0 @@
-# This file has runs the normal tests and also adds tests that can only be run
-# locally on a machine with a sound card. It's mostly to put the library through
-# its paces assuming a human is listening.
-
-include("runtests.jl")
-
-# these default values are specific to my machines
-if Sys.iswindows()
-    default_indev = "Microphone Array (Realtek High "
-    default_outdev = "Speaker/Headphone (Realtek High"
-elseif Sys.isapple()
-    default_indev = "Built-in Microphone"
-    default_outdev = "Built-in Output"
-elseif Sys.islinux()
-    default_indev = "default"
-    default_outdev = "default"
-end
-
-@testset "Local Tests" begin
-    @testset "Open Default Device" begin
-        println("Recording...")
-        stream = PortAudioStream(2, 0)
-        buf = read(stream, 5s)
-        close(stream)
-        @test size(buf) == (round(Int, 5 * samplerate(stream)), nchannels(stream.source))
-        println("Playing back recording...")
-        stream = PortAudioStream(0, 2)
-        write(stream, buf)
-        println("flushing...")
-        flush(stream)
-        close(stream)
-        println("Testing pass-through")
-        stream = PortAudioStream(2, 2)
-        write(stream, stream, 5s)
-        flush(stream)
-        close(stream)
-        println("done")
-    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)
-        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))
-        write(stream, buf)
-        io = IOBuffer()
-        show(io, stream)
-        @test occursin(
-            """
-PortAudioStream{Float32}
-  Samplerate: 44100.0Hz
-  Buffer Size: 4096 frames
-  2 channel sink: "$default_outdev"
-  2 channel source: "$default_indev\"""",
-            String(take!(io)),
-        )
-        close(stream)
-    end
-    @testset "Error on wrong name" begin
-        @test_throws ErrorException PortAudioStream("foobarbaz")
-    end
-    # 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
-        stream = PortAudioStream(0, 2)
-        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
-        @test fetch(t2) == 48000
-        flush(stream)
-        close(stream)
-    end
-    @testset "Queued Reading" begin
-        stream = PortAudioStream(2, 0)
-        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
-        @test fetch(t2) == 48000
-        close(stream)
-    end
-end
				`@ -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.