zymon/PortAudio.jl
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eeked some more allocation out of SinOsc

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This commit is contained in:
Spencer Russell 2014-06-25 22:08:03 -04:00
parent 7aafc697b0
commit 6a065f0af2
3 changed files with 24 additions and 24 deletions
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7
src/nodes.jl
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@ -40,13 +40,12 @@ function render(node::SinOscRenderer{Float32}, device_input::AudioBuf,
freq = node.freq freq = node.freq
# make sure these are Float32s so that we don't allocate doing conversions # make sure these are Float32s so that we don't allocate doing conversions
# in the tight loop # in the tight loop
pii::Float32 = pi pi2::Float32 = 2pi
dt::Float32 = 1/info.sample_rate phase_inc::Float32 = 2pi * freq / info.sample_rate
i::Int = 1 i::Int = 1
while i <= info.buf_size while i <= info.buf_size
outbuf[i] = sin(phase) outbuf[i] = sin(phase)
phase += 2pii*freq*dt phase = (phase + phase_inc) % pi2
phase = phase % 2pii
i += 1 i += 1
end end
node.phase = phase node.phase = phase

19
test/test_AudioIO.jl
View file

@ -94,22 +94,3 @@ end
info("Testing Audio Device Listing...") info("Testing Audio Device Listing...")
d_list = get_audio_devices() d_list = get_audio_devices()
@test length(d_list) > 0 @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

22
test/test_nodes.jl
View file

@ -68,11 +68,31 @@ render_output = render(osc, dev_input, test_info)
render_output = render(osc, dev_input, test_info) render_output = render(osc, dev_input, test_info)
@test mse(render_output, @test mse(render_output,
test_vect[test_info.buf_size+1:2*test_info.buf_size]) < FLOAT_THRESH 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) stop(osc)
render_output = render(osc, dev_input, test_info) render_output = render(osc, dev_input, test_info)
@test render_output == AudioSample[] @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...") info("Testing ArrayPlayer...")
v = rand(AudioSample, 44100) v = rand(AudioSample, 44100)
player = ArrayPlayer(v) player = ArrayPlayer(v)