130 lines
4.5 KiB
Julia
130 lines
4.5 KiB
Julia
using Base.Test
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using AudioIO
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import AudioIO.AudioSample
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import AudioIO.AudioBuf
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import AudioIO.AudioRenderer
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import AudioIO.AudioNode
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import AudioIO.DeviceInfo
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import AudioIO.render
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# A TestNode just renders out 1:buf_size each frame
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type TestRenderer <: AudioRenderer end
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typealias TestNode AudioNode{TestRenderer}
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TestNode() = TestNode(TestRenderer())
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function render(node::TestRenderer,
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device_input::AudioBuf,
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info::DeviceInfo)
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return AudioSample[1:info.buf_size]
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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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#### 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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testnode = TestNode()
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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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test1 = TestNode()
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test2 = TestNode()
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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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# 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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@test render_output == AudioSample[]
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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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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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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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@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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println("alloc: ", @allocated render(osc, dev_input, test_info))
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@test 500 > (@allocated render(osc, dev_input, test_info))
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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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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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# give a vector just a bit larger than 1 buffer size
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v = rand(AudioSample, test_info.buf_size + 1)
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player = ArrayPlayer(v)
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render(player, dev_input, test_info)
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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:end]
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info("Testing Gain...")
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gained = TestNode() * 0.75
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render_output = render(gained, dev_input, test_info)
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@test render_output == 0.75 * AudioSample[1:test_info.buf_size]
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info("Testing LinRamp...")
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ramp = LinRamp(0.25, 0.80, 1)
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expected = convert(AudioBuf, linspace(0.25, 0.80, test_info.sample_rate+1))
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render_output = render(ramp, 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(ramp, 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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@test 300 > (@allocated render(ramp, dev_input, test_info))
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