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

.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

@@ -1,7 +1,12 @@
 *.swp
 *.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

.travis.yml

@@ -1,14 +0,0 @@
-# Documentation: http://docs.travis-ci.com/user/languages/julia/
-language: julia
-os:
-  - linux
-  - osx
-sudo: required
-julia:
-  - 0.6
-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'

Project.toml

@@ -0,0 +1,26 @@
+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"]

README.md

@@ -1,18 +1,23 @@
 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.
+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, blocksize=4096, 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
@@ -26,51 +31,77 @@ 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
 
 ```julia
 stream = PortAudioStream(2, 2)
-write(stream, stream)
+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
 ```
 
 ### Open your built-in microphone and speaker by name
 ```julia
-stream = PortAudioStream("Built-in Microph", "Built-in Output")
-write(stream, stream)
+PortAudioStream("default", "default") do stream
+    write(stream, stream)
+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}}
@@ -78,16 +109,18 @@ julia> buf = read(stream, 10s)
 ▁▄▂▃▅▃▂▄▃▂▂▁▁▂▂▁▁▄▃▁▁▄▂▁▁▁▄▃▁▁▃▃▁▁▁▁▁▁▁▁▄▄▄▄▄▂▂▂▁▃▃▁▃▄▂▁▁▁▁▃▃▂▁▁▁▁▁▁▃▃▂▂▁▃▃▃▁▁▁▁
 ▁▄▂▃▅▃▂▄▃▂▂▁▁▂▂▁▁▄▃▁▁▄▂▁▁▁▄▃▁▁▃▃▁▁▁▁▁▁▁▁▄▄▄▄▄▂▂▂▁▃▃▁▃▄▂▁▁▁▁▃▃▂▁▁▁▁▁▁▃▃▂▂▁▃▃▃▁▁▁▁
 
+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
+```

REQUIRE

@@ -1,6 +0,0 @@
-julia 0.6-
-BinDeps
-SampledSignals 0.3.0
-RingBuffers 1.0.0
-@osx Homebrew
-@windows WinRPM

appveyor.yml

@@ -1,27 +0,0 @@
-environment:
-  matrix:
-  - JULIAVERSION: "julialang/bin/winnt/x64/0.6/julia-0.6-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"

deps/build.jl

@@ -1,26 +0,0 @@
-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, ))

deps/src/Makefile

@@ -1,68 +0,0 @@
-# Makefile lifted from Clang.jl
-# Copyright (c) 2012-: Isaiah Norton and [contributors](https://github.com/ihnorton/Clang.jl/graphs/contributors)
-
-ifeq (exists, $(shell [ -e Make.user ] && echo exists ))
-include Make.user
-endif
-
-TARGETDIR = ../usr/lib
-OBJS = pa_shim.o
-
-# Figure out OS and architecture
-OS=$(shell uname)
-ifneq (,$(findstring MINGW,$(OS)))
-  OS=WINNT
-endif
-
-# file extensions and platform-specific libs
-ifeq ($(OS), WINNT)
-  LIBS +=
-  LDFLAGS += -shared -L../../../RingBuffers/deps/usr/lib -lpa_ringbuffer
-  INC +=
-  SHACMD = sha256sum
-  SHLIB_EXT = dll
-else ifeq ($(OS), Darwin)
-  LIBS +=
-  INC +=
-# we'll rely on Julia to load RingBuffers.jl, which will in turn load the C
-# library that we depend on for these symbols
-  LDFLAGS += -dynamiclib -Wl,-undefined,dynamic_lookup
-  SHLIB_EXT = dylib
-  SHACMD = shasum -a256
-else
-  LIBS +=
-  INC +=
-  LDFLAGS += -shared
-  SHLIB_EXT = so
-  SHACMD = sha256sum
-endif
-
-SOURCEHASH = $(shell $(SHACMD) pa_shim.c | awk '{print $$1}')
-
-CFLAGS += -I../../../RingBuffers/deps/src -Wall -Wno-strict-aliasing -fno-omit-frame-pointer -fPIC -DSOURCEHASH=\"$(SOURCEHASH)\"
-LDFLAGS +=
-# LINUX_LIBS =-lrt
-# LINUX_LDFLAGS =-rdynamic
-# add the Homebrew.jl tree to the include dirs in case we used it for
-# portaudio and libsndfile
-# DARWIN_LDFLAGS =-L../../../Homebrew/deps/usr/lib
-# DARWIN_INC =-I../../../Homebrew/deps/usr/include
-
-TARGET = $(TARGETDIR)/pa_shim.$(SHLIB_EXT)
-
-.PHONY: clean default
-
-default: $(TARGET)
-
-%.o: %.c Makefile
-	$(CC) $< -fPIC -c -o $@ $(INC) $(CFLAGS)
-
-$(TARGETDIR):
-	mkdir -p $@
-
-$(TARGET): $(OBJS) $(TARGETDIR) Makefile
-	$(CC) $(OBJS) -shared -o $@ $(LDFLAGS) $(LIBS)
-
-clean:
-	rm -f $(OBJS)
-	rm -f $(TARGET)

deps/src/pa_shim.c

@@ -1,104 +0,0 @@
-#include <portaudio.h>
-#include <pa_ringbuffer.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <string.h>
-
-#define MIN(x, y) ((x) < (y) ? (x) : (y))
-
-typedef enum {
-    PA_SHIM_ERRMSG_OVERFLOW, // input overflow
-    PA_SHIM_ERRMSG_UNDERFLOW, // output underflow
-    PA_SHIM_ERRMSG_ERR_OVERFLOW, // error buffer overflowed
-} pa_shim_errmsg_t;
-
-// this callback type is used to notify the Julia side that the portaudio
-// callback has run
-typedef void (*pa_shim_notifycb_t)(void *userdata);
-
-// This struct is shared between the Julia side and C
-typedef struct {
-    PaUtilRingBuffer *inputbuf; // ringbuffer for input
-    PaUtilRingBuffer *outputbuf; // ringbuffer for output
-    PaUtilRingBuffer *errorbuf; // ringbuffer to send error notifications
-    int sync; // keep input/output ring buffers synchronized (0/1)
-    pa_shim_notifycb_t notifycb; // Julia callback to notify conditions
-    void *inputhandle; // condition to notify on new input
-    void *outputhandle; // condition to notify when ready for output
-    void *errorhandle; // condition to notify on new error
-} pa_shim_info_t;
-
-void senderr(pa_shim_info_t *info, pa_shim_errmsg_t msg) {
-    if(PaUtil_GetRingBufferWriteAvailable(info->errorbuf) < 2) {
-        // we've overflowed our error buffer! notify the host.
-        msg = PA_SHIM_ERRMSG_ERR_OVERFLOW;
-    }
-    PaUtil_WriteRingBuffer(info->errorbuf, &msg, 1);
-    if(info->notifycb) {
-        info->notifycb(info->errorhandle);
-    }
-}
-
-// return the sha256 hash of the shim source so we can make sure things are in sync
-const char *pa_shim_getsourcehash(void)
-{
-    // defined on the command-line at build-time
-    return SOURCEHASH;
-}
-
-/*
- * This routine will be called by the PortAudio engine when audio is needed.
- * It may called at interrupt level on some machines so don't do anything that
- * could mess up the system like calling malloc() or free().
- */
-int pa_shim_processcb(const void *input, void *output,
-                     unsigned long frameCount,
-                     const PaStreamCallbackTimeInfo* timeInfo,
-                     PaStreamCallbackFlags statusFlags,
-                     void *userData)
-{
-    pa_shim_info_t *info = userData;
-    if(info->notifycb == NULL) {
-        fprintf(stderr, "pa_shim ERROR: notifycb is NULL\n");
-    }
-    int nwrite;
-    if(info->inputbuf) {
-        nwrite = PaUtil_GetRingBufferWriteAvailable(info->inputbuf);
-        nwrite = MIN(frameCount, nwrite);
-    }
-    int nread;
-    if(info->outputbuf) {
-        nread = PaUtil_GetRingBufferReadAvailable(info->outputbuf);
-        nread = MIN(frameCount, nread);
-    }
-    if(info->inputbuf && info->outputbuf && info->sync) {
-        // to keep the buffers synchronized, set readable and writable to
-        // their minimum value
-        nread = MIN(nread, nwrite);
-        nwrite = nread;
-    }
-    // read/write from the ringbuffers
-    if(info->inputbuf) {
-        PaUtil_WriteRingBuffer(info->inputbuf, input, nwrite);
-        if(info->notifycb) {
-            info->notifycb(info->inputhandle);
-        }
-        if(nwrite < frameCount) {
-            senderr(info, PA_SHIM_ERRMSG_OVERFLOW);
-        }
-    }
-    if(info->outputbuf) {
-        PaUtil_ReadRingBuffer(info->outputbuf, output, nread);
-        if(info->notifycb) {
-            info->notifycb(info->outputhandle);
-        }
-        if(nread < frameCount) {
-            senderr(info, PA_SHIM_ERRMSG_UNDERFLOW);
-            // we didn't fill the whole output buffer, so zero it out
-            memset(output+nread*info->outputbuf->elementSizeBytes, 0,
-                   (frameCount - nread)*info->outputbuf->elementSizeBytes);
-        }
-    }
-
-    return paContinue;
-}

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

@@ -1,43 +1,49 @@
 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; blocksize=512)
+    mic = PortAudioStream(1, 0; latency = 0.1)
 
     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
-        print_with_color(barcolor(metersize, position), ">")
+        printstyled(">", color = barcolor(metersize, position))
     end
 
-    print(" " ^ blankchars)
-    print_with_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

@@ -0,0 +1,65 @@
+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

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

@@ -3,10 +3,10 @@
 
 module SpectrumExample
 
-using GR, PortAudio, SampledSignals
+using GR, PortAudio, SampledSignals, FFTW
 
 const N = 1024
-const stream = PortAudioStream(1, 0, blocksize=N)
+const stream = PortAudioStream(1, 0)
 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

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

@@ -0,0 +1,67 @@
+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

examples/waterfall_lines.jl

@@ -0,0 +1,43 @@
+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

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

@@ -1,250 +1,439 @@
-# Low-level wrappers for Portaudio calls
+module LibPortAudio
+
+using libportaudio_jll
+export libportaudio_jll
+
+function Pa_GetVersion()
+    ccall((:Pa_GetVersion, libportaudio), Cint, ())
+end
+
+function Pa_GetVersionText()
+    ccall((:Pa_GetVersionText, libportaudio), Ptr{Cchar}, ())
+end
+
+mutable struct PaVersionInfo
+    versionMajor::Cint
+    versionMinor::Cint
+    versionSubMinor::Cint
+    versionControlRevision::Ptr{Cchar}
+    versionText::Ptr{Cchar}
+end
+
+# no prototype is found for this function at portaudio.h:114:22, please use with caution
+function Pa_GetVersionInfo()
+    ccall((:Pa_GetVersionInfo, libportaudio), Ptr{PaVersionInfo}, ())
+end
 
-# General type aliases
-const PaTime = Cdouble
 const PaError = Cint
-const 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{Void}
-const PaStreamCallback = Void
-const PaStreamFlags = Culong
 
-const paNoFlag = PaStreamFlags(0x00)
+@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
 
-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)
+function Pa_GetErrorText(errorCode)
+    ccall((:Pa_GetErrorText, libportaudio), Ptr{Cchar}, (PaError,), errorCode)
+end
 
 function Pa_Initialize()
-    err = ccall((:Pa_Initialize, libportaudio), PaError, ())
-    handle_status(err)
+    ccall((:Pa_Initialize, libportaudio), PaError, ())
 end
 
 function Pa_Terminate()
-    err = ccall((:Pa_Terminate, libportaudio), PaError, ())
-    handle_status(err)
+    ccall((:Pa_Terminate, libportaudio), PaError, ())
 end
 
-Pa_GetVersion() = ccall((:Pa_GetVersion, libportaudio), Cint, ())
+const PaDeviceIndex = Cint
 
-function Pa_GetVersionText()
-    versionPtr = 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.
+@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
 
-# 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"
-)
-
-type PaHostApiInfo
-    struct_version::Cint
-    api_type::PaHostApiTypeId
+mutable struct PaHostApiInfo
+    structVersion::Cint
+    type::PaHostApiTypeId
     name::Ptr{Cchar}
     deviceCount::Cint
     defaultInputDevice::PaDeviceIndex
     defaultOutputDevice::PaDeviceIndex
 end
 
-Pa_GetHostApiInfo(i) = unsafe_load(ccall((:Pa_GetHostApiInfo, libportaudio),
-                                   Ptr{PaHostApiInfo}, (PaHostApiIndex,), i))
-
-# Device Functions
-
-type 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() = ccall((:Pa_GetDeviceCount, libportaudio), PaDeviceIndex, ())
+function Pa_HostApiTypeIdToHostApiIndex(type)
+    ccall(
+        (:Pa_HostApiTypeIdToHostApiIndex, libportaudio),
+        PaHostApiIndex,
+        (PaHostApiTypeId,),
+        type,
+    )
+end
 
-Pa_GetDeviceInfo(i) = unsafe_load(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() = ccall((:Pa_GetDefaultInputDevice, libportaudio),
-                                   PaDeviceIndex, ())
+mutable struct PaHostErrorInfo
+    hostApiType::PaHostApiTypeId
+    errorCode::Clong
+    errorText::Ptr{Cchar}
+end
 
-Pa_GetDefaultOutputDevice() = 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
 
-type 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{Void}
+    hostApiSpecificStreamInfo::Ptr{Cvoid}
 end
 
-type PaStreamInfo
+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
     structVersion::Cint
     inputLatency::PaTime
     outputLatency::PaTime
     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{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[]
+function Pa_GetStreamInfo(stream)
+    ccall((:Pa_GetStreamInfo, libportaudio), Ptr{PaStreamInfo}, (Ptr{PaStream},), stream)
 end
 
-function Pa_StartStream(stream::PaStream)
-    err = 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 = 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 = 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 = 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 = 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=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
+function Pa_GetStreamWriteAvailable(stream)
+    ccall((:Pa_GetStreamWriteAvailable, libportaudio), Clong, (Ptr{PaStream},), stream)
 end
 
-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
+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 = ccall((:Pa_GetErrorText, libportaudio),
-                        Ptr{Cchar}, (PaError,), err)
-            warn("libportaudio: " * unsafe_string(msg))
-        end
-    elseif err != PA_NO_ERROR
-        msg = ccall((:Pa_GetErrorText, libportaudio),
-                    Ptr{Cchar}, (PaError,), err)
-        error("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/pa_shim.jl

@@ -1,40 +0,0 @@
-function init_pa_shim()
-    global const libpa_shim = Libdl.find_library(
-            ["pa_shim"],
-            [joinpath(dirname(@__FILE__), "..", "deps", "usr", "lib")])
-    shim_dlib = Libdl.dlopen(libpa_shim)
-    # pointer to the shim's process callback
-    global const shim_processcb_c = Libdl.dlsym(shim_dlib, :pa_shim_processcb)
-    if shim_processcb_c == C_NULL
-        error("Got NULL pointer loading `pa_shim_processcb`")
-    end
-end
-
-const pa_shim_errmsg_t = Cint
-const PA_SHIM_ERRMSG_OVERFLOW = Cint(0) # input overflow
-const PA_SHIM_ERRMSG_UNDERFLOW = Cint(1) # output underflow
-const PA_SHIM_ERRMSG_ERR_OVERFLOW = Cint(2) # error buffer overflowed
-
-
-# This struct is shared with pa_shim.c
-mutable struct pa_shim_info_t
-    inputbuf::Ptr{PaUtilRingBuffer} # ringbuffer for input
-    outputbuf::Ptr{PaUtilRingBuffer} # ringbuffer for output
-    errorbuf::Ptr{PaUtilRingBuffer} # ringbuffer to send error notifications
-    sync::Cint # keep input/output ring buffers synchronized (0/1)
-    notifycb::Ptr{Void} # Julia callback to notify on updates (called from audio thread)
-    inputhandle::Ptr{Void} # condition to notify on new input data
-    outputhandle::Ptr{Void} # condition to notify when ready for output
-    errorhandle::Ptr{Void} # condition to notify on new errors
-end
-
-"""
-    PortAudio.shimhash()
-
-Return the sha256 hash(as a string) of the source file used to build the shim.
-We may use this sometime to verify that the distributed binary stays in sync
-with the rest of the package.
-"""
-shimhash() = unsafe_string(
-        ccall((:pa_shim_getsourcehash, libpa_shim), Cstring, ()))
-Base.unsafe_convert(::Type{Ptr{Void}}, info::pa_shim_info_t) = pointer_from_objref(info)

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))
+
+
+
+
+
+

src/suppressor.jl

@@ -1,20 +0,0 @@
-# while waiting for this PR to get merged: https://github.com/Ismael-VC/Suppressor.jl/pull/12
-# we'll just include the relevant code here
-
-macro suppress_err(block)
-    quote
-        if ccall(:jl_generating_output, Cint, ()) == 0
-            ORIGINAL_STDERR = STDERR
-            err_rd, err_wr = redirect_stderr()
-            err_reader = @async readstring(err_rd)
-        end
-
-        value = $(esc(block))
-
-        if ccall(:jl_generating_output, Cint, ()) == 0
-            redirect_stderr(ORIGINAL_STDERR)
-            close(err_wr)
-        end
-        value
-    end
-end

test/REQUIRE

@@ -1 +0,0 @@
-BaseTestNext

test/runtests.jl

@@ -1,333 +1,256 @@
 #!/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 Base.Test
-using TestSetExtensions
-using PortAudio
-using SampledSignals
-using RingBuffers
-
-# pull in some extra stuff we need to test the callback directly
-using PortAudio: notifyhandle, notifycb_c, shim_processcb_c
-using PortAudio: pa_shim_errmsg_t, pa_shim_info_t
-using PortAudio: PA_SHIM_ERRMSG_ERR_OVERFLOW, PA_SHIM_ERRMSG_UNDERFLOW, PA_SHIM_ERRMSG_OVERFLOW
-
-"Setup buffers to test callback behavior"
-function setup_callback(inchans, outchans, nframes, synced)
-    sourcebuf = RingBuffer{Float32}(inchans, nframes*2) # the microphone input should end up here
-    sinkbuf = RingBuffer{Float32}(outchans, nframes*2) # the callback should copy this to cb_output
-    errbuf = RingBuffer{pa_shim_errmsg_t}(1, 8)
-
-    # pass NULL for i/o we're not using
-    info = pa_shim_info_t(
-        inchans > 0 ? pointer(sourcebuf) : C_NULL,
-        outchans > 0 ? pointer(sinkbuf) : C_NULL,
-        pointer(errbuf),
-        synced, notifycb_c,
-        inchans > 0 ? notifyhandle(sourcebuf) : C_NULL,
-        outchans > 0 ? notifyhandle(sinkbuf) : C_NULL,
-        notifyhandle(errbuf)
-    )
-    flags = Culong(0)
-
-    cb_input = rand(Float32, inchans, nframes) # simulate microphone input
-    cb_output = rand(Float32, outchans, nframes) # this is where the output should go
-
-    function processfunc()
-        ccall(shim_processcb_c, Cint,
-            (Ptr{Float32}, Ptr{Float32}, Culong, Ptr{Void}, Culong, Ptr{Void}),
-            cb_input, cb_output, nframes, C_NULL, flags, pointer_from_objref(info))
-    end
-
-    (sourcebuf, sinkbuf, errbuf, cb_input, cb_output, processfunc)
-end
-
-function test_callback(inchans, outchans, synced)
-    nframes = 8
-    (sourcebuf, sinkbuf, errbuf,
-     cb_input, cb_output, process) = setup_callback(inchans, outchans,
-                                                    nframes, synced)
-    if outchans > 0
-        testout = rand(Float32, outchans, nframes) # generate some test data to play
-        write(sinkbuf, testout) # fill the output ringbuffer
-    end
-    @test process() == PortAudio.paContinue
-    if outchans > 0
-        # testout -> sinkbuf -> cb_output
-        @test cb_output == testout
-    end
-    if inchans > 0
-        # cb_input -> sourcebuf
-        @test read(sourcebuf, nframes) == cb_input
-    end
-    @test framesreadable(errbuf) == 0
-end
-
-"""
-    test_callback_underflow(inchans, outchans; nframes=8, underfill=3, synced=false)
-
-Test that the callback works on underflow conditions. underfill is the numer of
-frames we feed in, which should be less than nframes.
-"""
-function test_callback_underflow(inchans, outchans, synced)
-    nframes = 8
-    underfill = 3 # must be less than nframes
-    (sourcebuf, sinkbuf, errbuf,
-     cb_input, cb_output, process) = setup_callback(inchans, outchans,
-                                                    nframes, synced)
-    outchans > 0 || error("Can't test underflow with no output")
-    testout = rand(Float32, outchans, underfill)
-    write(sinkbuf, testout) # underfill the output ringbuffer
-    # call callback (partial underflow)
-    @test process() == PortAudio.paContinue
-    @test cb_output[:, 1:underfill] == testout
-    @test cb_output[:, (underfill+1):nframes] == zeros(Float32, outchans, (nframes-underfill))
-    errs = readavailable(errbuf)
-    if inchans > 0
-        received = readavailable(sourcebuf)
-        if synced
-            @test size(received, 2) == underfill
-            @test received == cb_input[:, 1:underfill]
-            @test length(errs) == 2
-            @test Set(errs) == Set([PA_SHIM_ERRMSG_UNDERFLOW, PA_SHIM_ERRMSG_OVERFLOW])
-        else
-            @test size(received, 2) == nframes
-            @test received == cb_input
-            @test length(errs) == 1
-            @test errs[1] == PA_SHIM_ERRMSG_UNDERFLOW
-        end
-    else
-        @test length(errs) == 1
-        @test errs[1] == PA_SHIM_ERRMSG_UNDERFLOW
-    end
-
-    # call again (total underflow)
-    @test process() == PortAudio.paContinue
-    @test cb_output == zeros(Float32, outchans, nframes)
-    errs = readavailable(errbuf)
-    if inchans > 0
-        received = readavailable(sourcebuf)
-        if synced
-            @test size(received, 2) == 0
-            @test length(errs) == 2
-            @test Set(errs) == Set([PA_SHIM_ERRMSG_UNDERFLOW, PA_SHIM_ERRMSG_OVERFLOW])
-        else
-            @test size(received, 2) == nframes
-            @test received == cb_input
-            @test length(errs) == 1
-            @test errs[1] == PA_SHIM_ERRMSG_UNDERFLOW
-        end
-    else
-        @test length(errs) == 1
-        @test errs[1] == PA_SHIM_ERRMSG_UNDERFLOW
-    end
-end
-
-function test_callback_overflow(inchans, outchans, synced)
-    nframes = 8
-    (sourcebuf, sinkbuf, errbuf,
-     cb_input, cb_output, process) = setup_callback(inchans, outchans,
-                                                    nframes, synced)
-    inchans > 0 || error("Can't test overflow with no input")
-    @test frameswritable(sinkbuf) == nframes*2
-
-    # the first time it should half-fill the input ring buffer
-    if outchans > 0
-        testout = rand(Float32, outchans, nframes)
-        write(sinkbuf, testout)
-    end
-    @test framesreadable(sourcebuf) == 0
-    outchans > 0 && @test frameswritable(sinkbuf) == nframes
-    @test process() == PortAudio.paContinue
-    @test framesreadable(errbuf) == 0
-    @test framesreadable(sourcebuf) == nframes
-    outchans > 0 && @test frameswritable(sinkbuf) == nframes*2
-
-    # now run the process func again to completely fill the input ring buffer
-    outchans > 0 && write(sinkbuf, testout)
-    @test framesreadable(sourcebuf) == nframes
-    outchans > 0 && @test frameswritable(sinkbuf) == nframes
-    @test process() == PortAudio.paContinue
-    @test framesreadable(errbuf) == 0
-    @test framesreadable(sourcebuf) == nframes*2
-    outchans > 0 && @test frameswritable(sinkbuf) == nframes*2
-
-    # now this time the process func should overflow the input buffer
-    outchans > 0 && write(sinkbuf, testout)
-    @test framesreadable(sourcebuf) == nframes*2
-    outchans > 0 && @test frameswritable(sinkbuf) == nframes
-    @test process() == PortAudio.paContinue
-    @test framesreadable(sourcebuf) == nframes*2
-    errs = readavailable(errbuf)
-    if outchans > 0
-        if synced
-            # if input and output are synced, thec callback didn't pull from
-            # the output ringbuf
-            @test frameswritable(sinkbuf) == nframes
-            @test cb_output == zeros(Float32, outchans, nframes)
-            @test length(errs) == 2
-            @test Set(errs) == Set([PA_SHIM_ERRMSG_UNDERFLOW, PA_SHIM_ERRMSG_OVERFLOW])
-        else
-            @test frameswritable(sinkbuf) == nframes*2
-            @test length(errs) == 1
-            @test errs[1] == PA_SHIM_ERRMSG_OVERFLOW
-        end
-    else
-        @test length(errs) == 1
-        @test errs[1] == PA_SHIM_ERRMSG_OVERFLOW
-    end
-end
-
-# these test are currently set up to run on OSX
-
-@testset DottedTestSet "PortAudio Tests" begin
-    if is_windows()
-        default_indev = "Microphone Array (Realtek High "
-        default_outdev = "Speaker/Headphone (Realtek High"
-    elseif is_apple()
-        default_indev = "Built-in Microph"
-        default_outdev = "Built-in Output"
-    elseif is_linux()
-        default_indev = "default"
-        default_outdev = "default"
-    end
-
-    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 "Tests without sound" begin
     @testset "Reports version" begin
         io = IOBuffer()
         PortAudio.versioninfo(io)
         result = split(String(take!((io))), "\n")
         # make sure this is the same version I tested with
         @test startswith(result[1], "PortAudio V19")
-        @test result[3] == "Shim Source Hash: 4ea2a8526b"
     end
 
-    @testset "Basic callback functionality" begin
-        @testset "basic duplex (no sync)" begin
-            test_callback(2, 3, false)
-        end
-        @testset "basic input-only (no sync)" begin
-            test_callback(2, 0, false)
-        end
-        @testset "basic output-only (no sync)" begin
-            test_callback(0, 2, false)
-        end
-        @testset "basic no input or output (no sync)" begin
-            test_callback(0, 0, false)
-        end
-        @testset "basic duplex (sync)" begin
-            test_callback(2, 3, true)
-        end
-        @testset "basic input-only (sync)" begin
-            test_callback(2, 0, true)
-        end
-        @testset "basic output-only (sync)" begin
-            test_callback(0, 2, true)
-        end
-        @testset "basic no input or output (sync)" begin
-            test_callback(0, 0, true)
-        end
+    @testset "Can list devices without crashing" begin
+        display(devices())
+        println()
     end
 
-    @testset "Ouput underflow" begin
-        @testset "underflow duplex (nosync)" begin
-            test_callback_underflow(2, 3, false)
-        end
-        @testset "underflow output-only (nosync)" begin
-            test_callback_underflow(0, 3, false)
-        end
-        @testset "underflow duplex (sync)" begin
-            test_callback_underflow(2, 3, true)
-        end
-        @testset "underflow output-only (sync)" begin
-            test_callback_underflow(0, 3, true)
+    @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 "Input overflow" begin
-        @testset "overflow duplex (nosync)" begin
-            test_callback_overflow(2, 3, false)
-        end
-        @testset "overflow input-only (nosync)" begin
-            test_callback_overflow(2, 0, false)
-        end
-        @testset "overflow duplex (sync)" begin
-            test_callback_overflow(2, 3, true)
-        end
-        @testset "overflow input-only (sync)" begin
-            test_callback_overflow(2, 0, true)
-        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
 
-    @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(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 String(take!(io)) == """
-        PortAudio.PortAudioStream{Float32}
-          Samplerate: 44100.0Hz
-          Buffer Size: 4096 frames
-          2 channel sink: "$default_outdev"
-          2 channel source: "$default_indev\""""
-        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
+    # 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

@@ -0,0 +1,95 @@
+# 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