more fixes to get the tests running without deprecations on 0.4 and 0.3

.travis.yml

@@ -1,21 +1,21 @@
-language: cpp
+language: julia
 compiler:
   - clang
 notifications:
   email: spencer.f.russell@gmail.com
-env:
+julia:
-    - JULIAVERSION="julianightlies"
+  - 0.3
-    - JULIAVERSION="juliareleases"
+  - 0.4
-before_install:
+#  - nightly
-  - sudo add-apt-repository ppa:staticfloat/julia-deps -y
+# Fails with:
-  - sudo add-apt-repository ppa:staticfloat/$JULIAVERSION -y
+# LoadError: ccall: could not find function jl_read_sonames
-  - sudo apt-get update -qq -y
+# while loading /home/travis/.julia/v0.5/AudioIO/deps/build.jl, in expression starting on line 29
-  - sudo apt-get install libpcre3-dev julia -y
-  - if [[ -a .git/shallow ]]; then git fetch --unshallow; fi
 script:
+  - if [[ -a .git/shallow ]]; then git fetch --unshallow; fi
   - julia -e 'Pkg.init()'
   - julia -e 'Pkg.add("BinDeps"); Pkg.checkout("BinDeps")' # latest master needed for Pacman support
   - julia -e 'Pkg.clone(pwd()); Pkg.build("AudioIO")'
-  - julia -e 'Pkg.test("AudioIO", coverage=true)'
+  - julia -e 'Pkg.add("FactCheck")' # add FactCheck manually because we're not using Pkg.test()
+  - julia --code-coverage=user test/runtests.jl # Pkg.test disables inlining when enabling coverage, which kills our allocation tests
 after_success:
 - if [ $JULIAVERSION = "juliareleases" ]; then julia -e 'cd(Pkg.dir("AudioIO")); Pkg.add("Coverage"); using Coverage; Coveralls.submit(Coveralls.process_folder())'; fi

deps/build.jl

@@ -1,4 +1,5 @@
 using BinDeps
+using Compat
 
 @BinDeps.setup
 
@@ -26,5 +27,5 @@ end
     provides(WinRPM.RPM, "libsndfile1", libsndfile, os = :Windows)
 end
 
-@BinDeps.install [:libportaudio => :libportaudio,
+@BinDeps.install @compat(Dict(:libportaudio => :libportaudio,
-                  :libsndfile => :libsndfile]
+                  :libsndfile => :libsndfile))

src/nodes.jl

@@ -228,7 +228,7 @@ function play(arr::AudioBuf, args...)
 end
 
 # If the array is the wrong floating type, convert it
-@compat function play{T <: AbstractFloat}(arr::Array{T}, args...)
+function play{T <: AbstractFloat}(arr::Array{T}, args...)
     arr = convert(AudioBuf, arr)
     play(arr, args...)
 end
@@ -306,7 +306,7 @@ export LinRamp
 
 function render(node::LinRampRenderer, device_input::AudioBuf, info::DeviceInfo)
     # Resize buffer if (1) it's too small or (2) we've hit the end of the ramp
-    ramp_samples::Int = int(node.duration * info.sample_rate)
+    ramp_samples::Int = round(Int, node.duration * info.sample_rate)
     block_samples = min(ramp_samples, info.buf_size)
     if length(node.buf) != block_samples
         resize!(node.buf, block_samples)

src/portaudio.jl

@@ -23,7 +23,7 @@ const paInt8    = convert(PaSampleFormat, 0x10)
 const paUInt8   = convert(PaSampleFormat, 0x20)
 
 # PaHostApiTypeId values
-@compat const pa_host_api_names = (
+@compat const pa_host_api_names = Dict(
     0 => "In Development", # use while developing support for a new host API
     1 => "Direct Sound",
     2 => "MME",
@@ -338,7 +338,7 @@ type PaHostApiInfo
     defaultOutputDevice::PaDeviceIndex
 end
 
-@compat type PortAudioInterface <: AudioInterface
+type PortAudioInterface <: AudioInterface
     name::AbstractString
     host_api::AbstractString
     max_input_channels::Int

src/sndfile.jl

@@ -16,7 +16,7 @@ const SF_SEEK_SET = 0
 const SF_SEEK_CUR = 1
 const SF_SEEK_END = 2
 
-@compat const EXT_TO_FORMAT = (
+@compat const EXT_TO_FORMAT = Dict(
     ".wav" => SF_FORMAT_WAV,
     ".flac" => SF_FORMAT_FLAC
 )
@@ -28,12 +28,6 @@ type SF_INFO
     format::Int32
     sections::Int32
     seekable::Int32
-
-    function SF_INFO(frames::Integer, samplerate::Integer, channels::Integer,
-                     format::Integer, sections::Integer, seekable::Integer)
-        new(int64(frames), int32(samplerate), int32(channels), int32(format),
-            int32(sections), int32(seekable))
-    end
 end
 
 type AudioFile
@@ -45,7 +39,7 @@ samplerate(f::AudioFile) = f.sfinfo.samplerate
 
 # AudioIO.open is part of the public API, but is not exported so that it
 # doesn't conflict with Base.open
-@compat function open(path::AbstractString, mode::AbstractString = "r",
+function open(path::AbstractString, mode::AbstractString = "r",
             sampleRate::Integer = 44100, channels::Integer = 1,
             format::Integer = 0)
     @assert channels <= 2
@@ -66,11 +60,11 @@ samplerate(f::AudioFile) = f.sfinfo.samplerate
     end
 
     filePtr = ccall((:sf_open, libsndfile), Ptr{Void},
-                    (Ptr{Uint8}, Int32, Ptr{SF_INFO}),
+                    (Ptr{UInt8}, Int32, Ptr{SF_INFO}),
                     path, file_mode, &sfinfo)
 
     if filePtr == C_NULL
-        errmsg = ccall((:sf_strerror, libsndfile), Ptr{Uint8}, (Ptr{Void},), filePtr)
+        errmsg = ccall((:sf_strerror, libsndfile), Ptr{UInt8}, (Ptr{Void},), filePtr)
         error(bytestring(errmsg))
     end
 
@@ -132,7 +126,7 @@ Base.read(file::AudioFile) = Base.read(file, Int16)
 
 function Base.write{T}(file::AudioFile, frames::Array{T})
     @assert file.sfinfo.channels <= 2
-    nframes = int(length(frames) / file.sfinfo.channels)
+    nframes = round(Int, length(frames) / file.sfinfo.channels)
 
     if T == Int16
         return ccall((:sf_writef_short, libsndfile), Int64,
@@ -180,7 +174,7 @@ end
 
 typealias FilePlayer AudioNode{FileRenderer}
 FilePlayer(file::AudioFile) = FilePlayer(FileRenderer(file))
-@compat FilePlayer(path::AbstractString) = FilePlayer(AudioIO.open(path))
+FilePlayer(path::AbstractString) = FilePlayer(AudioIO.open(path))
 
 function render(node::FileRenderer, device_input::AudioBuf, info::DeviceInfo)
     @assert node.file.sfinfo.samplerate == info.sample_rate
@@ -204,7 +198,7 @@ function render(node::FileRenderer, device_input::AudioBuf, info::DeviceInfo)
     end
 end
 
-@compat function play(filename::AbstractString, args...)
+function play(filename::AbstractString, args...)
     player = FilePlayer(filename)
     play(player, args...)
 end

test/test_AudioIO.jl

@@ -1,6 +1,7 @@
 module TestAudioIO
 
 using FactCheck
+using Compat
 using AudioIO
 import AudioIO.AudioBuf
 
@@ -43,32 +44,32 @@ facts("Array playback") do
         f32 = convert(Array{Float32}, sin(phase))
         test_stream = TestAudioStream()
         player = play(f32, test_stream)
-        @fact process(test_stream) => f32[1:TEST_BUF_SIZE]
+        @fact process(test_stream) --> f32[1:TEST_BUF_SIZE]
     end
 
     context("Playing Float64 arrays") do
         f64 = convert(Array{Float64}, sin(phase))
         test_stream = TestAudioStream()
         player = play(f64, test_stream)
-        @fact process(test_stream) => convert(AudioBuf, f64[1:TEST_BUF_SIZE])
+        @fact process(test_stream) --> convert(AudioBuf, f64[1:TEST_BUF_SIZE])
     end
 
     context("Playing Int8(Signed) arrays") do
-        i8 = Int8[-127:127]
+        i8 = Int8[-127:127;]
         test_stream = TestAudioStream()
         player = play(i8, test_stream)
-        @fact process(test_stream)[1:255] =>
+        @fact process(test_stream)[1:255] -->
-                mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
+                mse(convert(AudioBuf, collect(linspace(-1.0, 1.0, 255))))
     end
 
-    context("Playing Uint8(Unsigned) arrays") do
+    context("Playing UInt8(Unsigned) arrays") do
         # for unsigned 8-bit audio silence is represented as 128, so the symmetric range
         # is 1-255
-        ui8 = Uint8[1:255]
+        ui8 = UInt8[1:255;]
         test_stream = TestAudioStream()
         player = play(ui8, test_stream)
-        @fact process(test_stream)[1:255] =>
+        @fact process(test_stream)[1:255] -->
-                mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
+                mse(convert(AudioBuf, collect(linspace(-1.0, 1.0, 255))))
    end
 end
 
@@ -78,12 +79,12 @@ facts("AudioNode Stopping") do
     play(node, test_stream)
     process(test_stream)
     stop(node)
-    @fact process(test_stream) => zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
+    @fact process(test_stream) --> zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
 end
 
 facts("Audio Device Listing") do
     # there aren't any devices on the Travis machine so just test that this doesn't crash
-    @fact get_audio_devices() => issubtype(Array)
+    @fact get_audio_devices() --> issubtype(Array)
 end
 
 end # module TestAudioIO

test/test_nodes.jl

@@ -1,5 +1,6 @@
 module TestAudioIONodes
 
+using Compat
 using FactCheck
 using AudioIO
 import AudioIO: AudioSample, AudioBuf, AudioRenderer, AudioNode
@@ -10,7 +11,7 @@ include("testhelpers.jl")
 # A TestNode just renders out 1:buf_size each frame
 type TestRenderer <: AudioRenderer
     buf::AudioBuf
-    TestRenderer(buf_size::Integer) = new(AudioSample[1:buf_size])
+    TestRenderer(buf_size::Integer) = new(AudioSample[1:buf_size;])
 end
 
 typealias TestNode AudioNode{TestRenderer}
@@ -31,7 +32,7 @@ facts("Validating TestNode allocation") do
     test = TestNode(test_info.buf_size)
     # JIT
     render(test, dev_input, test_info)
-    @fact (@allocated render(test, dev_input, test_info)) => 16
+    @fact (@allocated render(test, dev_input, test_info)) --> 16
 end
 
 #### AudioMixer Tests ####
@@ -43,16 +44,17 @@ facts("AudioMixer") do
     context("0 Input Mixer") do
         mix = AudioMixer()
         render_output = render(mix, dev_input, test_info)
-        @fact render_output => AudioSample[]
+        @fact render_output --> AudioSample[]
-        @fact (@allocated render(mix, dev_input, test_info)) => 48
+        # 0.4 uses 64 bytes, 0.3 uses 48
+        @fact (@allocated render(mix, dev_input, test_info)) --> less_than(65)
     end
 
     context("1 Input Mixer") do
         testnode = TestNode(test_info.buf_size)
         mix = AudioMixer([testnode])
         render_output = render(mix, dev_input, test_info)
-        @fact render_output => AudioSample[1:test_info.buf_size]
+        @fact render_output --> AudioSample[1:test_info.buf_size;]
-        @fact (@allocated render(mix, dev_input, test_info)) => 64
+        @fact (@allocated render(mix, dev_input, test_info)) --> 64
     end
 
     context("2 Input Mixer") do
@@ -61,17 +63,17 @@ facts("AudioMixer") do
         mix = AudioMixer([test1, test2])
         render_output = render(mix, dev_input, test_info)
         # make sure the two inputs are being added together
-        @fact render_output => 2 * AudioSample[1:test_info.buf_size]
+        @fact render_output --> 2 * AudioSample[1:test_info.buf_size;]
-        @fact (@allocated render(mix, dev_input, test_info)) => 96
+        @fact (@allocated render(mix, dev_input, test_info)) --> 96
         # now we'll stop one of the inputs and make sure it gets removed
         stop(test1)
         render_output = render(mix, dev_input, test_info)
         # make sure the two inputs are being added together
-        @fact render_output => AudioSample[1:test_info.buf_size]
+        @fact render_output --> AudioSample[1:test_info.buf_size;]
 
         stop(mix)
         render_output = render(mix, dev_input, test_info)
-        @fact render_output => AudioSample[]
+        @fact render_output --> AudioSample[]
     end
 end
 
@@ -81,25 +83,25 @@ facts("SinOSC") do
     freq = 440
     # note that this range includes the end, which is why there are
     # sample_rate+1 samples
-    t = linspace(0, 1, int(test_info.sample_rate+1))
+    t = linspace(0, 1, round(Int, test_info.sample_rate+1))
     test_vect = convert(AudioBuf, sin(2pi * t * freq))
     context("Fixed Frequency") do
         osc = SinOsc(freq)
         render_output = render(osc, dev_input, test_info)
-        @fact mse(render_output, test_vect[1:test_info.buf_size]) =>
+        @fact mse(render_output, test_vect[1:test_info.buf_size]) -->
                 lessthan(MSE_THRESH)
         render_output = render(osc, dev_input, test_info)
         @fact mse(render_output,
-                test_vect[test_info.buf_size+1:2*test_info.buf_size]) =>
+                test_vect[test_info.buf_size+1:2*test_info.buf_size]) -->
                 lessthan(MSE_THRESH)
-        @fact (@allocated render(osc, dev_input, test_info)) => 64
+        @fact (@allocated render(osc, dev_input, test_info)) --> 64
         stop(osc)
         render_output = render(osc, dev_input, test_info)
-        @fact render_output => AudioSample[]
+        @fact render_output --> AudioSample[]
     end
 
     context("Testing SinOsc with signal input") do
-        t = linspace(0, 1, int(test_info.sample_rate+1))
+        t = linspace(0, 1, round(Int, test_info.sample_rate+1))
         f = 440 .- t .* (440-110)
         dt = 1 / test_info.sample_rate
         # NOTE - this treats the phase as constant at each sample, which isn't strictly
@@ -112,14 +114,14 @@ facts("SinOSC") do
         freq = LinRamp(440, 110, 1)
         osc = SinOsc(freq)
         render_output = render(osc, dev_input, test_info)
-        @fact mse(render_output, expected[1:test_info.buf_size]) =>
+        @fact mse(render_output, expected[1:test_info.buf_size]) -->
                 lessthan(MSE_THRESH)
         render_output = render(osc, dev_input, test_info)
         @fact mse(render_output,
-                expected[test_info.buf_size+1:2*test_info.buf_size]) =>
+                expected[test_info.buf_size+1:2*test_info.buf_size]) -->
                 lessthan(MSE_THRESH)
         # give a bigger budget here because we're rendering 2 nodes
-        @fact (@allocated render(osc, dev_input, test_info)) => 160
+        @fact (@allocated render(osc, dev_input, test_info)) --> 160
     end
 end
 
@@ -127,7 +129,7 @@ facts("AudioInput") do
     node = AudioInput()
     test_data = rand(AudioSample, test_info.buf_size)
     render_output = render(node, test_data, test_info)
-    @fact render_output => test_data
+    @fact render_output --> test_data
 end
 
 facts("ArrayPlayer") do
@@ -135,13 +137,13 @@ facts("ArrayPlayer") do
         v = rand(AudioSample, 44100)
         player = ArrayPlayer(v)
         render_output = render(player, dev_input, test_info)
-        @fact render_output => v[1:test_info.buf_size]
+        @fact render_output --> v[1:test_info.buf_size]
         render_output = render(player, dev_input, test_info)
-        @fact render_output => v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
+        @fact render_output --> v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
-        @fact (@allocated render(player, dev_input, test_info)) => 192
+        @fact (@allocated render(player, dev_input, test_info)) --> 192
         stop(player)
         render_output = render(player, dev_input, test_info)
-        @fact render_output => AudioSample[]
+        @fact render_output --> AudioSample[]
     end
 
     context("testing end of vector") do
@@ -150,7 +152,7 @@ facts("ArrayPlayer") do
         player = ArrayPlayer(v)
         render(player, dev_input, test_info)
         render_output = render(player, dev_input, test_info)
-        @fact render_output => v[test_info.buf_size+1:end]
+        @fact render_output --> v[test_info.buf_size+1:end]
     end
 end
 
@@ -158,35 +160,35 @@ facts("Gain") do
     context("Constant Gain") do
         gained = TestNode(test_info.buf_size) * 0.75
         render_output = render(gained, dev_input, test_info)
-        @fact render_output => 0.75 * AudioSample[1:test_info.buf_size]
+        @fact render_output --> 0.75 * AudioSample[1:test_info.buf_size;]
-        @fact (@allocated render(gained, dev_input, test_info)) => 32
+        @fact (@allocated render(gained, dev_input, test_info)) --> 32
     end
     context("Gain by a Signal") do
         gained = TestNode(test_info.buf_size) * TestNode(test_info.buf_size)
         render_output = render(gained, dev_input, test_info)
-        @fact render_output => AudioSample[1:test_info.buf_size] .* AudioSample[1:test_info.buf_size]
+        @fact render_output --> AudioSample[1:test_info.buf_size;] .* AudioSample[1:test_info.buf_size;]
-        @fact (@allocated render(gained, dev_input, test_info)) => 48
+        @fact (@allocated render(gained, dev_input, test_info)) --> 48
     end
 end
 
 facts("LinRamp") do
     ramp = LinRamp(0.25, 0.80, 1)
-    expected = convert(AudioBuf, linspace(0.25, 0.80, int(test_info.sample_rate+1)))
+    expected = convert(AudioBuf, collect(linspace(0.25, 0.80, round(Int, test_info.sample_rate+1))))
     render_output = render(ramp, dev_input, test_info)
-    @fact mse(render_output, expected[1:test_info.buf_size]) =>
+    @fact mse(render_output, expected[1:test_info.buf_size]) -->
             lessthan(MSE_THRESH)
     render_output = render(ramp, dev_input, test_info)
     @fact mse(render_output,
-            expected[(test_info.buf_size+1):(2*test_info.buf_size)]) =>
+            expected[(test_info.buf_size+1):(2*test_info.buf_size)]) -->
             lessthan(MSE_THRESH)
-    @fact (@allocated render(ramp, dev_input, test_info)) => 64
+    @fact (@allocated render(ramp, dev_input, test_info)) --> 64
 end
 
 facts("Offset") do
     offs = TestNode(test_info.buf_size) + 0.5
     render_output = render(offs, dev_input, test_info)
-    @fact render_output => 0.5 + AudioSample[1:test_info.buf_size]
+    @fact render_output --> 0.5 + AudioSample[1:test_info.buf_size;]
-    @fact (@allocated render(offs, dev_input, test_info)) => 32
+    @fact (@allocated render(offs, dev_input, test_info)) --> 32
 end
 
 end # module TestAudioIONodes

test/test_sndfile.jl

@@ -1,19 +1,20 @@
 module TestSndfile
 
-include("testhelpers.jl")
+using Compat
-
-using AudioIO
 using FactCheck
+using AudioIO
 import AudioIO: DeviceInfo, render, AudioSample, AudioBuf
 
+include("testhelpers.jl")
+
 facts("WAV file write/read") do
     fname = Pkg.dir("AudioIO", "test", "sinwave.wav")
 
     srate = 44100
     freq = 440
-    t = [0 : 2 * srate - 1] / srate
+    t = collect(0 : 2 * srate - 1) / srate
     phase = 2 * pi * freq * t
-    reference = int16((2 ^ 15 - 1) * sin(phase))
+    reference = round(Int16, (2 ^ 15 - 1) * sin(phase))
 
     AudioIO.open(fname, "w") do f
         write(f, reference)
@@ -21,12 +22,12 @@ facts("WAV file write/read") do
 
     # test basic reading
     AudioIO.open(fname) do f
-        @fact f.sfinfo.channels => 1
+        @fact f.sfinfo.channels --> 1
-        @fact f.sfinfo.frames => 2 * srate
+        @fact f.sfinfo.frames --> 2 * srate
         actual = read(f)
-        @fact length(reference) => length(actual)
+        @fact length(reference) --> length(actual)
-        @fact reference => actual[:, 1]
+        @fact reference --> actual[:, 1]
-        @fact samplerate(f) => srate
+        @fact samplerate(f) --> srate
     end
 
     # test seeking
@@ -41,21 +42,21 @@ facts("WAV file write/read") do
         # convert to floating point because that's what AudioIO uses natively
         expected = convert(AudioBuf, reference ./ (2^15))
         buf = render(node, input, test_info)
-        @fact expected[1:bufsize] => buf[1:bufsize]
+        @fact expected[1:bufsize] --> buf[1:bufsize]
         buf = render(node, input, test_info)
-        @fact expected[bufsize+1:2*bufsize] => buf[1:bufsize]
+        @fact expected[bufsize+1:2*bufsize] --> buf[1:bufsize]
     end
 end
 
 facts("Stereo file reading") do
     fname = Pkg.dir("AudioIO", "test", "440left_880right.wav")
     srate = 44100
-    t = [0 : 2 * srate - 1] / srate
+    t = collect(0:(2 * srate - 1)) / srate
-    expected = int16((2^15-1) * hcat(sin(2pi*t*440), sin(2pi*t*880)))
+    expected = round(Int16, (2^15-1) * hcat(sin(2pi*t*440), sin(2pi*t*880)))
 
     AudioIO.open(fname) do f
         buf = read(f)
-        @fact buf => mse(expected, 5)
+        @fact buf --> mse(expected, 5)
     end
 end
 
@@ -67,15 +68,15 @@ facts("Stereo file rendering") do
     bufsize = 1024
     input = zeros(AudioSample, bufsize)
     test_info = DeviceInfo(srate, bufsize)
-    t = [0 : 2 * srate - 1] / srate
+    t = collect(0 : 2 * srate - 1) / srate
     expected = convert(AudioBuf, 0.5 * (sin(2pi*t*440) + sin(2pi*t*880)))
 
     AudioIO.open(fname) do f
         node = FilePlayer(f)
         buf = render(node, input, test_info)
-        @fact buf[1:bufsize] => mse(expected[1:bufsize])
+        @fact buf[1:bufsize] --> mse(expected[1:bufsize])
         buf = render(node, input, test_info)
-        @fact buf[1:bufsize] => mse(expected[bufsize+1:2*bufsize])
+        @fact buf[1:bufsize] --> mse(expected[bufsize+1:2*bufsize])
     end
 end