using Base.Test using AudioIO import AudioIO.AudioSample import AudioIO.AudioBuf import AudioIO.AudioRenderer import AudioIO.AudioNode import AudioIO.DeviceInfo import AudioIO.render # A TestNode just renders out 1:buf_size each frame type TestRenderer <: AudioRenderer end typealias TestNode AudioNode{TestRenderer} TestNode() = TestNode(TestRenderer()) function render(node::TestRenderer, device_input::AudioBuf, info::DeviceInfo) return AudioSample[1:info.buf_size] end test_info = DeviceInfo(44100, 512) dev_input = zeros(AudioSample, test_info.buf_size) #### AudioMixer Tests #### # TODO: there should be a setup/teardown mechanism and some way to isolate # tests info("Testing AudioMixer...") mix = AudioMixer() render_output = render(mix, dev_input, test_info) @test render_output == AudioSample[] testnode = TestNode() mix = AudioMixer([testnode]) render_output = render(mix, dev_input, test_info) @test render_output == AudioSample[1:test_info.buf_size] test1 = TestNode() test2 = TestNode() mix = AudioMixer([test1, test2]) render_output = render(mix, dev_input, test_info) # make sure the two inputs are being added together @test render_output == 2 * AudioSample[1:test_info.buf_size] # 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 @test render_output == AudioSample[1:test_info.buf_size] stop(mix) render_output = render(mix, dev_input, test_info) @test render_output == AudioSample[] # TODO: I think we can do better than this const FLOAT_THRESH = 1e-9 info("Testing SinOSC...") freq = 440 # note that this range includes the end, which is why there are sample_rate+1 samples t = linspace(0, 1, test_info.sample_rate+1) test_vect = convert(AudioBuf, sin(2pi * t * freq)) osc = SinOsc(freq) render_output = render(osc, dev_input, test_info) @test mse(render_output, test_vect[1:test_info.buf_size]) < FLOAT_THRESH render_output = render(osc, dev_input, test_info) @test mse(render_output, test_vect[test_info.buf_size+1:2*test_info.buf_size]) < FLOAT_THRESH @test 600 > (@allocated render(osc, dev_input, test_info)) stop(osc) render_output = render(osc, dev_input, test_info) @test render_output == AudioSample[] info("Testing ArrayPlayer...") v = rand(AudioSample, 44100) player = ArrayPlayer(v) render_output = render(player, dev_input, test_info) @test render_output == v[1:test_info.buf_size] render_output = render(player, dev_input, test_info) @test render_output == v[(test_info.buf_size + 1) : (2*test_info.buf_size)] stop(player) render_output = render(player, dev_input, test_info) @test render_output == AudioSample[] # 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] info("Testing Gain...") gained = TestNode() * 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]) < 1e-16 render_output = render(ramp, dev_input, test_info) @test mse(render_output, expected[(test_info.buf_size+1):(2*test_info.buf_size)]) < 1e-14