ports tests over to RunTest.jl
This commit is contained in:
Spencer Russell
parent
201d2852c6
commit
389636fda2
6 changed files with 253 additions and 184 deletions
1
test/REQUIRE
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1
test/REQUIRE
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@ -0,0 +1 @@
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FactCheck
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18
test/runtests.jl
Executable file
18
test/runtests.jl
Executable file
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@ -0,0 +1,18 @@
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#!/usr/bin/env julia
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using FactCheck
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test_regex = r"^test_.*\.jl$"
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test_dir = "test"
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test_files = filter(n -> ismatch(test_regex, n), readdir(test_dir))
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if length(test_files) == 0
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error("No test files found. Make sure you're running from the root directory")
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end
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for test_file in test_files
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include(test_file)
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end
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# return the overall exit status
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exitstatus()
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28
test/test.jl
28
test/test.jl
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@ -1,28 +0,0 @@
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#!/usr/bin/env julia
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test_regex = r"^test_.*\.jl$"
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test_dir = "test"
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test_files = filter(n -> ismatch(test_regex, n), readdir(test_dir))
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if length(test_files) == 0
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error("No test files found. Make sure you're running from the root directory")
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end
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# convenience function to calculate the mean-squared error
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function mse(arr1::Array, arr2::Array)
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@assert length(arr1) == length(arr2)
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N = length(arr1)
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err = 0.0
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for i in 1:N
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err += (arr2[i] - arr1[i])^2
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end
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err /= N
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end
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for test_file in test_files
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info("")
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info("Running tests from \"$(test_file)\"...")
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info("===================================================================")
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include(test_file)
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info("===================================================================")
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end
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@ -1,10 +1,33 @@
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using Base.Test
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module TestAudioIO
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using FactCheck
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using AudioIO
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import AudioIO.AudioBuf
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const TEST_SAMPLERATE = 44100
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const TEST_BUF_SIZE = 1024
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include("testhelpers.jl")
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# convenience function to calculate the mean-squared error
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function mse(arr1::Array, arr2::Array)
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@assert length(arr1) == length(arr2)
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N = length(arr1)
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err = 0.0
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for i in 1:N
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err += (arr2[i] - arr1[i])^2
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end
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err /= N
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end
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mse(X::AbstractArray, thresh=1e-8) = Y::AbstractArray -> begin
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if size(X) != size(Y)
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return false
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end
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return mse(X, Y) < thresh
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end
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type TestAudioStream <: AudioIO.AudioStream
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root::AudioIO.AudioMixer
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info::AudioIO.DeviceInfo
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@ -28,69 +51,78 @@ end
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#### Test playing back various vector types ####
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# data shared between tests, for convenience
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t = linspace(0, 2, 2 * 44100)
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phase = 2pi * 100 * t
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facts("Array playback") do
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# data shared between tests, for convenience
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t = linspace(0, 2, 2 * 44100)
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phase = 2pi * 100 * t
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## Test Float32 arrays, this is currently the native audio playback format
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info("Testing Playing Float32 arrays...")
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f32 = convert(Array{Float32}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f32, test_stream)
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@test process(test_stream) == f32[1:TEST_BUF_SIZE]
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## Test Float32 arrays, this is currently the native audio playback format
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context("Playing Float32 arrays") do
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f32 = convert(Array{Float32}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f32, test_stream)
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@fact process(test_stream) => f32[1:TEST_BUF_SIZE]
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end
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context("Playing Float64 arrays") do
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f64 = convert(Array{Float64}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f64, test_stream)
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@fact process(test_stream) => convert(AudioBuf, f64[1:TEST_BUF_SIZE])
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end
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info("Testing Playing Float64 arrays...")
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f64 = convert(Array{Float64}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f64, test_stream)
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@test process(test_stream) == convert(AudioBuf, f64[1:TEST_BUF_SIZE])
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context("Playing Int8(Signed) arrays") do
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i8 = Int8[-127:127]
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test_stream = TestAudioStream()
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player = play(i8, test_stream)
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@fact process(test_stream)[1:255] =>
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mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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end
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info("Testing Playing Int8(Signed) arrays...")
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i8 = Int8[-127:127]
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test_stream = TestAudioStream()
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player = play(i8, test_stream)
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@test_approx_eq(process(test_stream)[1:255],
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convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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info("Testing Playing Uint8(Unsigned) arrays...")
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# for unsigned 8-bit audio silence is represented as 128, so the symmetric range
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# is 1-255
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ui8 = Uint8[1:255]
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test_stream = TestAudioStream()
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player = play(ui8, test_stream)
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@test_approx_eq(process(test_stream)[1:255],
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convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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info("Testing AudioNode Stopping...")
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test_stream = TestAudioStream()
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node = SinOsc(440)
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play(node, test_stream)
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process(test_stream)
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stop(node)
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@test process(test_stream) == zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
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info("Testing wav file write/read")
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fname = "test/sinwave.wav"
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samplerate = 44100
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freq = 440
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t = [0 : 2 * samplerate - 1] / samplerate
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phase = 2 * pi * freq * t
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reference = int16((2 ^ 15 - 1) * sin(phase))
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af_open(fname, "w") do f
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write(f, reference)
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context("Playing Uint8(Unsigned) arrays") do
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# for unsigned 8-bit audio silence is represented as 128, so the symmetric range
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# is 1-255
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ui8 = Uint8[1:255]
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test_stream = TestAudioStream()
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player = play(ui8, test_stream)
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@fact process(test_stream)[1:255] =>
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mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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end
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end
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af_open(fname) do f
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@test f.sfinfo.channels == 1
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@test f.sfinfo.frames == 2 * samplerate
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actual = read(f, 2 * samplerate)
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@test_approx_eq(reference, actual)
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facts("AudioNode Stopping") do
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test_stream = TestAudioStream()
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node = SinOsc(440)
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play(node, test_stream)
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process(test_stream)
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stop(node)
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@fact process(test_stream) => zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
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end
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info("Testing Audio Device Listing...")
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# there aren't any devices on the Travis machine so just test that this doesn't crash
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get_audio_devices()
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facts("WAV file write/read") do
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fname = "test/sinwave.wav"
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samplerate = 44100
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freq = 440
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t = [0 : 2 * samplerate - 1] / samplerate
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phase = 2 * pi * freq * t
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reference = int16((2 ^ 15 - 1) * sin(phase))
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af_open(fname, "w") do f
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write(f, reference)
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end
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af_open(fname) do f
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@fact f.sfinfo.channels => 1
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@fact f.sfinfo.frames => 2 * samplerate
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actual = read(f, 2 * samplerate)
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@fact reference => mse(actual)
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end
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end
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facts("Audio Device Listing") do
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# there aren't any devices on the Travis machine so just test that this doesn't crash
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@fact get_audio_devices() => issubtype(Array)
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end
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end # module TestAudioIO
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@ -1,4 +1,6 @@
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using Base.Test
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module TestAudioIONodes
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using FactCheck
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using AudioIO
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import AudioIO.AudioSample
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import AudioIO.AudioBuf
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@ -7,6 +9,8 @@ import AudioIO.AudioNode
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import AudioIO.DeviceInfo
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import AudioIO.render
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include("testhelpers.jl")
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# A TestNode just renders out 1:buf_size each frame
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type TestRenderer <: AudioRenderer
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buf::AudioBuf
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@ -25,117 +29,145 @@ end
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test_info = DeviceInfo(44100, 512)
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dev_input = zeros(AudioSample, test_info.buf_size)
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# first validate that the TestNode doesn't allocate so it doesn't mess up our
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# other tests
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test = TestNode(test_info.buf_size)
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# JIT
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render(test, dev_input, test_info)
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@test ((@allocated render(test, dev_input, test_info)) < 20)
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facts("Validating TestNode allocation") do
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# first validate that the TestNode doesn't allocate so it doesn't mess up our
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# other tests
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test = TestNode(test_info.buf_size)
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# JIT
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render(test, dev_input, test_info)
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@fact (@allocated render(test, dev_input, test_info)) => lessthan(20)
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end
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#### AudioMixer Tests ####
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# TODO: there should be a setup/teardown mechanism and some way to isolate
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# tests
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info("Testing AudioMixer...")
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mix = AudioMixer()
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render_output = render(mix, dev_input, test_info)
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@test render_output == AudioSample[]
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facts("AudioMixer") do
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context("0 Input Mixer") do
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mix = AudioMixer()
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[]
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@fact (@allocated render(mix, dev_input, test_info)) => lessthan(49)
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end
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testnode = TestNode(test_info.buf_size)
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mix = AudioMixer([testnode])
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render_output = render(mix, dev_input, test_info)
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@test render_output == AudioSample[1:test_info.buf_size]
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@test 100 > (@allocated render(mix, dev_input, test_info))
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context("1 Input Mixer") do
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testnode = TestNode(test_info.buf_size)
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mix = AudioMixer([testnode])
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[1:test_info.buf_size]
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@fact (@allocated render(mix, dev_input, test_info)) => lessthan(65)
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end
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test1 = TestNode(test_info.buf_size)
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test2 = TestNode(test_info.buf_size)
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mix = AudioMixer([test1, test2])
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@test render_output == 2 * AudioSample[1:test_info.buf_size]
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context("2 Input Mixer") do
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test1 = TestNode(test_info.buf_size)
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test2 = TestNode(test_info.buf_size)
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mix = AudioMixer([test1, test2])
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@fact render_output => 2 * AudioSample[1:test_info.buf_size]
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@fact (@allocated render(mix, dev_input, test_info)) => lessthan(97)
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# now we'll stop one of the inputs and make sure it gets removed
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stop(test1)
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@fact render_output => AudioSample[1:test_info.buf_size]
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# now we'll stop one of the inputs and make sure it gets removed
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stop(test1)
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@test render_output == AudioSample[1:test_info.buf_size]
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stop(mix)
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[]
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end
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end
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stop(mix)
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render_output = render(mix, dev_input, test_info)
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@test render_output == AudioSample[]
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MSE_THRESH = 1e-7
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# TODO: I think we can do better than this
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const MSE_THRESH = 1e-7
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facts("SinOSC") do
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freq = 440
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# note that this range includes the end, which is why there are
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# sample_rate+1 samples
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t = linspace(0, 1, test_info.sample_rate+1)
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test_vect = convert(AudioBuf, sin(2pi * t * freq))
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context("Fixed Frequency") do
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osc = SinOsc(freq)
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render_output = render(osc, dev_input, test_info)
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@fact mse(render_output, test_vect[1:test_info.buf_size]) =>
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lessthan(MSE_THRESH)
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render_output = render(osc, dev_input, test_info)
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@fact mse(render_output,
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test_vect[test_info.buf_size+1:2*test_info.buf_size]) =>
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lessthan(MSE_THRESH)
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@fact (@allocated render(osc, dev_input, test_info)) => lessthan(200)
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stop(osc)
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render_output = render(osc, dev_input, test_info)
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@fact render_output => AudioSample[]
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end
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info("Testing SinOSC...")
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freq = 440
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# note that this range includes the end, which is why there are sample_rate+1 samples
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t = linspace(0, 1, test_info.sample_rate+1)
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test_vect = convert(AudioBuf, sin(2pi * t * freq))
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osc = SinOsc(freq)
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render_output = render(osc, dev_input, test_info)
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@test mse(render_output, test_vect[1:test_info.buf_size]) < MSE_THRESH
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render_output = render(osc, dev_input, test_info)
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@test mse(render_output,
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test_vect[test_info.buf_size+1:2*test_info.buf_size]) < MSE_THRESH
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@test 200 > (@allocated render(osc, dev_input, test_info))
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stop(osc)
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render_output = render(osc, dev_input, test_info)
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@test render_output == AudioSample[]
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context("Testing SinOsc with signal input") do
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t = linspace(0, 1, test_info.sample_rate+1)
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f = 440 .- t .* (440-110)
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dt = 1 / test_info.sample_rate
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# NOTE - this treats the phase as constant at each sample, which isn't strictly
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# true. Unfortunately doing this correctly requires knowing more about the
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# modulating signal and doing the real integral
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phase = cumsum(2pi * dt .* f)
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unshift!(phase, 0)
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expected = convert(AudioBuf, sin(phase))
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info("Testing SinOsc with signal input")
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t = linspace(0, 1, test_info.sample_rate+1)
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f = 440 .- t .* (440-110)
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dt = 1 / test_info.sample_rate
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# NOTE - this treats the phase as constant at each sample, which isn't strictly
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# true. Unfortunately doing this correctly requires knowing more about the
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# modulating signal and doing the real integral
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phase = cumsum(2pi * dt .* f)
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unshift!(phase, 0)
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expected = convert(AudioBuf, sin(phase))
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freq = LinRamp(440, 110, 1)
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osc = SinOsc(freq)
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render_output = render(osc, dev_input, test_info)
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@fact mse(render_output, expected[1:test_info.buf_size]) =>
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lessthan(MSE_THRESH)
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render_output = render(osc, dev_input, test_info)
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@fact mse(render_output,
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expected[test_info.buf_size+1:2*test_info.buf_size]) =>
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lessthan(MSE_THRESH)
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# give a bigger budget here because we're rendering 2 nodes
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@fact (@allocated render(osc, dev_input, test_info)) => lessthan(500)
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end
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end
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freq = LinRamp(440, 110, 1)
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osc = SinOsc(freq)
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render_output = render(osc, dev_input, test_info)
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@test mse(render_output, expected[1:test_info.buf_size]) < MSE_THRESH
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render_output = render(osc, dev_input, test_info)
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@test mse(render_output,
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expected[test_info.buf_size+1:2*test_info.buf_size]) < MSE_THRESH
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# give a bigger budget here because we're rendering 2 nodes
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@test 500 > (@allocated render(osc, dev_input, test_info))
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facts("ArrayPlayer") do
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context("playing long sample") do
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v = rand(AudioSample, 44100)
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player = ArrayPlayer(v)
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render_output = render(player, dev_input, test_info)
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@fact render_output => v[1:test_info.buf_size]
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render_output = render(player, dev_input, test_info)
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@fact render_output => v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
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@fact (@allocated render(player, dev_input, test_info)) => lessthan(200)
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stop(player)
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render_output = render(player, dev_input, test_info)
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@fact render_output => AudioSample[]
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end
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info("Testing ArrayPlayer...")
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v = rand(AudioSample, 44100)
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player = ArrayPlayer(v)
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render_output = render(player, dev_input, test_info)
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@test render_output == v[1:test_info.buf_size]
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render_output = render(player, dev_input, test_info)
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@test render_output == v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
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@test 200 > (@allocated render(player, dev_input, test_info))
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stop(player)
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render_output = render(player, dev_input, test_info)
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@test render_output == AudioSample[]
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context("testing end of vector") do
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# give a vector just a bit larger than 1 buffer size
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v = rand(AudioSample, test_info.buf_size + 1)
|
||||
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]
|
||||
end
|
||||
end
|
||||
|
||||
# give a vector just a bit larger than 1 buffer size
|
||||
v = rand(AudioSample, test_info.buf_size + 1)
|
||||
player = ArrayPlayer(v)
|
||||
render(player, dev_input, test_info)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@test render_output == v[test_info.buf_size+1:end]
|
||||
facts("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]
|
||||
end
|
||||
|
||||
info("Testing Gain...")
|
||||
facts("LinRamp") do
|
||||
ramp = LinRamp(0.25, 0.80, 1)
|
||||
expected = convert(AudioBuf, linspace(0.25, 0.80, test_info.sample_rate+1))
|
||||
render_output = render(ramp, dev_input, test_info)
|
||||
@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)]) =>
|
||||
lessthan(MSE_THRESH)
|
||||
@fact (@allocated render(ramp, dev_input, test_info)) => lessthan(300)
|
||||
end
|
||||
|
||||
gained = TestNode(test_info.buf_size) * 0.75
|
||||
render_output = render(gained, dev_input, test_info)
|
||||
@test render_output == 0.75 * AudioSample[1:test_info.buf_size]
|
||||
|
||||
info("Testing LinRamp...")
|
||||
ramp = LinRamp(0.25, 0.80, 1)
|
||||
expected = convert(AudioBuf, linspace(0.25, 0.80, test_info.sample_rate+1))
|
||||
render_output = render(ramp, dev_input, test_info)
|
||||
@test mse(render_output, expected[1:test_info.buf_size]) < MSE_THRESH
|
||||
render_output = render(ramp, dev_input, test_info)
|
||||
@test mse(render_output,
|
||||
expected[(test_info.buf_size+1):(2*test_info.buf_size)]) < MSE_THRESH
|
||||
@test 300 > (@allocated render(ramp, dev_input, test_info))
|
||||
end # module TestAudioIONodes
|
||||
|
|
|
|||
14
test/testhelpers.jl
Normal file
14
test/testhelpers.jl
Normal file
|
|
@ -0,0 +1,14 @@
|
|||
# convenience function to calculate the mean-squared error
|
||||
function mse(arr1::Array, arr2::Array)
|
||||
@assert length(arr1) == length(arr2)
|
||||
N = length(arr1)
|
||||
err = 0.0
|
||||
for i in 1:N
|
||||
err += (arr2[i] - arr1[i])^2
|
||||
end
|
||||
err /= N
|
||||
end
|
||||
|
||||
issubtype(T::Type) = x -> typeof(x) <: T
|
||||
lessthan(rhs) = lhs -> lhs < rhs
|
||||
greaterthan(rhs) = lhs -> lhs > rhs
|
||||
Loading…
Reference in a new issue