zymon/PortAudio.jl
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got LinRamp allocation down to 240bytes

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This commit is contained in:
Spencer Russell 2014-06-25 22:25:35 -04:00
parent 6a065f0af2
commit e451509293
2 changed files with 39 additions and 26 deletions
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21
src/nodes.jl
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@ -213,6 +213,9 @@ type LinRampRenderer <: AudioRenderer
start::AudioSample
finish::AudioSample
dur::Float32
buf::AudioBuf
LinRampRenderer(start, finish, dur) = new(start, finish, dur, AudioSample[])
end
typealias LinRamp AudioNode{LinRampRenderer}
@ -225,13 +228,21 @@ export LinRamp
function render(node::LinRampRenderer, device_input::AudioBuf, info::DeviceInfo)
ramp_samples = int(node.dur * info.sample_rate)
block_samples = min(ramp_samples, info.buf_size)
out_block = Array(AudioSample, block_samples)
for i in 1:block_samples
out_block[i] = node.start + ((i-1) / ramp_samples) *
(node.finish - node.start)
if length(node.buf) != block_samples
resize!(node.buf, block_samples)
end
out_block = node.buf
i::Int = 1
val::AudioSample = node.start
inc::AudioSample = (node.finish - node.start) / ramp_samples
while i <= block_samples
out_block[i] = val
i += 1
val += inc
end
node.dur -= block_samples / info.sample_rate
node.start += block_samples / ramp_samples * (node.finish - node.start)
node.start = val
return out_block
end

44
test/test_nodes.jl
View file

@ -73,25 +73,25 @@ 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 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)
@ -121,6 +121,8 @@ 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
@test mse(render_output, expected[1:test_info.buf_size]) < FLOAT_THRESH
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
@test mse(render_output,
expected[(test_info.buf_size+1):(2*test_info.buf_size)]) < FLOAT_THRESH
@test 300 > (@allocated render(ramp, dev_input, test_info))