SinOsc{Float32} allocation is way down

src/nodes.jl

@@ -16,10 +16,11 @@ end
 
 type SinOscRenderer{T<:Union(Float32, AudioNode)} <: AudioRenderer
     freq::T
-    phase::Float64
+    phase::Float32
+    buf::AudioBuf
 
     function SinOscRenderer(freq)
-        new(freq, 0.0)
+        new(freq, 0.0, AudioSample[])
     end
 end
 
@@ -31,12 +32,22 @@ export SinOsc
 
 function render(node::SinOscRenderer{Float32}, device_input::AudioBuf,
         info::DeviceInfo)
-    outbuf = Array(AudioSample, info.buf_size)
+    if length(node.buf) != info.buf_size
+        resize!(node.buf, info.buf_size)
+    end
+    outbuf = node.buf
     phase = node.phase
-    dt = 1/info.sample_rate
+    freq = node.freq
-    for i in 1:info.buf_size
+    # make sure these are Float32s so that we don't allocate doing conversions
+    # in the tight loop
+    pii::Float32 = pi
+    dt::Float32 = 1/info.sample_rate
+    i::Int = 1
+    while i <= info.buf_size
         outbuf[i] = sin(phase)
-        phase += 2pi*node.freq*dt
+        phase += 2pii*freq*dt
+        phase = phase % 2pii
+        i += 1
     end
     node.phase = phase
     return outbuf

test/test_nodes.jl

@@ -7,9 +7,6 @@ import AudioIO.AudioNode
 import AudioIO.DeviceInfo
 import AudioIO.render
 
-test_info = DeviceInfo(44100, 512)
-dev_input = zeros(AudioSample, test_info.buf_size)
-
 # A TestNode just renders out 1:buf_size each frame
 type TestRenderer <: AudioRenderer end
 
@@ -22,6 +19,9 @@ function render(node::TestRenderer,
     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
@@ -54,6 +54,9 @@ 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
@@ -61,10 +64,11 @@ 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 render_output == test_vect[1:test_info.buf_size]
+@test mse(render_output, test_vect[1:test_info.buf_size]) < FLOAT_THRESH
 render_output = render(osc, dev_input, test_info)
-@test render_output == test_vect[test_info.buf_size+1:2*test_info.buf_size]
+@test mse(render_output,
-@test 50000 > (@allocated render(osc, dev_input, test_info))
+        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[]