From 6a065f0af2e64c3352c2f66e7b0b6559be3e562c Mon Sep 17 00:00:00 2001
From: Spencer Russell <spencer.f.russell@gmail.com>
Date: Wed, 25 Jun 2014 22:08:03 -0400
Subject: [PATCH] eeked some more allocation out of SinOsc

---
 src/nodes.jl         |  7 +++----
 test/test_AudioIO.jl | 19 -------------------
 test/test_nodes.jl   | 22 +++++++++++++++++++++-
 3 files changed, 24 insertions(+), 24 deletions(-)

diff --git a/src/nodes.jl b/src/nodes.jl
index d71d067..195096c 100644
--- a/src/nodes.jl
+++ b/src/nodes.jl
@@ -40,13 +40,12 @@ function render(node::SinOscRenderer{Float32}, device_input::AudioBuf,
     freq = node.freq
     # 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
+    pi2::Float32 = 2pi
+    phase_inc::Float32 = 2pi * freq / info.sample_rate
     i::Int = 1
     while i <= info.buf_size
         outbuf[i] = sin(phase)
-        phase += 2pii*freq*dt
-        phase = phase % 2pii
+        phase = (phase + phase_inc) % pi2
         i += 1
     end
     node.phase = phase
diff --git a/test/test_AudioIO.jl b/test/test_AudioIO.jl
index 0077f6e..1373c49 100644
--- a/test/test_AudioIO.jl
+++ b/test/test_AudioIO.jl
@@ -94,22 +94,3 @@ end
 info("Testing Audio Device Listing...")
 d_list = get_audio_devices()
 @test length(d_list) > 0
-
-info("Testing param control with signals")
-t = linspace(0, 1, TEST_SAMPLERATE+1)
-f = 440 .- t .* (440-110)
-dt = 1 / TEST_SAMPLERATE
-# NOTE - this treats the phase as constant at each sample, which isn't strictly
-# true. Unfortunately doing this correctly requires knowing more about the
-# modulating signal and doing the real integral
-phase = cumsum(2pi * dt .* f)
-unshift!(phase, 0)
-expected = convert(AudioBuf, sin(phase))
-
-test_stream = TestAudioStream()
-freq = LinRamp(440, 110, 1)
-player = play(SinOsc(freq), test_stream)
-out = process(test_stream)
-#println("expected: $(expected[1:30])")
-#println("got: $(out[1:30])")
-@test mse(out, expected[1:TEST_BUF_SIZE]) < 1e-16
diff --git a/test/test_nodes.jl b/test/test_nodes.jl
index f49f1da..5df01b6 100644
--- a/test/test_nodes.jl
+++ b/test/test_nodes.jl
@@ -68,11 +68,31 @@ render_output = render(osc, dev_input, test_info)
 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))
+@test 200 > (@allocated render(osc, dev_input, test_info))
 stop(osc)
 render_output = render(osc, dev_input, test_info)
 @test render_output == AudioSample[]
 
+info("Testing SinOsc with  signal input")
+t = linspace(0, 1, 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
+# true. Unfortunately doing this correctly requires knowing more about the
+# modulating signal and doing the real integral
+phase = cumsum(2pi * dt .* f)
+unshift!(phase, 0)
+expected = convert(AudioBuf, sin(phase))
+
+freq = LinRamp(440, 110, 1)
+osc = SinOsc(freq)
+render_output = render(osc, dev_input, test_info)
+@test mse(render_output, expected[1:test_info.buf_size]) < FLOAT_THRESH
+render_output = render(osc, dev_input, test_info)
+@test mse(render_output,
+        expected[test_info.buf_size+1:2*test_info.buf_size]) < FLOAT_THRESH
+#@test 400 > (@allocated render(osc, dev_input, test_info))
+
 info("Testing ArrayPlayer...")
 v = rand(AudioSample, 44100)
 player = ArrayPlayer(v)