refactored AudioNode to contain a AudioRenderer, tests passing
This commit is contained in:
Spencer Russell
parent
0ddd57c0a9
commit
a1ed357629
7 changed files with 211 additions and 219 deletions
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@ -12,11 +12,12 @@ typealias AudioSample Float32
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# A frame of audio, possibly multi-channel
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typealias AudioBuf Array{AudioSample}
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# A node in the render tree
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abstract AudioNode
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# used as a type parameter for AudioNodes. Subtypes handle the actual DSP for
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# each node
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abstract AudioRenderer
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# A stream of audio (for instance that writes to hardware)
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# All AudioStream subtypes should have a mixer and info field
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# A stream of audio (for instance that writes to hardware). All AudioStream
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# subtypes should have a mixer and info field
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abstract AudioStream
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# An audio interface is usually a physical sound card, but could
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@ -29,6 +30,31 @@ type DeviceInfo
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buf_size::Integer
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end
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type AudioNode{T<:AudioRenderer}
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active::Bool
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end_cond::Condition
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renderer::T
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function AudioNode(renderer::AudioRenderer)
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new(true, Condition(), renderer)
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end
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end
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function render(node::AudioNode, input::AudioBuf, info::DeviceInfo)
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# TODO: not sure if the compiler will infer that render() always returns an
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# AudioBuf. Might need to help it
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if node.active
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result = render(node.renderer, input, info)
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if length(result) < info.buf_size
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node.active = false
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notify(node.end_cond)
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end
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return result
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else
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return AudioSample[]
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end
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end
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include("nodes.jl")
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include("portaudio.jl")
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include("sndfile.jl")
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@ -38,8 +64,7 @@ include("operators.jl")
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# Play an AudioNode by adding it as an input to the root mixer node
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function play(node::AudioNode, stream::AudioStream)
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activate(node)
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add_input(stream.mixer, node)
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push!(stream.root, node)
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return node
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end
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@ -53,26 +78,13 @@ function play(node::AudioNode)
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end
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function stop(node::AudioNode)
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deactivate(node)
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node
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end
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function activate(node::AudioNode)
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node.active = true
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end
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function deactivate(node::AudioNode)
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node.active = false
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notify(node.deactivate_cond)
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end
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function is_active(node::AudioNode)
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node.active
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notify(node.end_cond)
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end
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function Base.wait(node::AudioNode)
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if is_active(node)
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wait(node.deactivate_cond)
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if node.active
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wait(node.end_cond)
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end
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end
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208
src/nodes.jl
208
src/nodes.jl
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@ -1,146 +1,120 @@
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#### NullNode ####
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type NullRenderer <: AudioRenderer end
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typealias NullNode AudioNode{NullRenderer}
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NullNode() = NullNode(NullRenderer())
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export NullNode
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function render(node::NullRenderer, device_input::AudioBuf, info::DeviceInfo)
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# TODO: preallocate buffer
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return zeros(info.buf_size)
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end
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#### SinOsc ####
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# Generates a sin tone at the given frequency
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export SinOsc
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type SinOsc <: AudioNode
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active::Bool
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deactivate_cond::Condition
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type SinOscRenderer <: AudioRenderer
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freq::Real
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phase::FloatingPoint
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function SinOsc(freq::Real)
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new(false, Condition(), freq, 0.0)
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function SinOscRenderer(freq::Real)
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new(freq, 0.0)
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end
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end
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function render(node::SinOsc, device_input::AudioBuf, info::DeviceInfo)
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phase = AudioSample[1:info.buf_size] * 2pi * node.freq / info.sample_rate
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typealias SinOsc AudioNode{SinOscRenderer}
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SinOsc(freq::Real) = SinOsc(SinOscRenderer(freq))
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SinOsc() = SinOsc(440)
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export SinOsc
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function render(node::SinOscRenderer, device_input::AudioBuf, info::DeviceInfo)
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phase = AudioSample[0:(info.buf_size-1)] * 2pi * node.freq / info.sample_rate
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phase .+= node.phase
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node.phase = phase[end]
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return sin(phase), is_active(node)
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node.phase = phase[end] + 2pi * node.freq / info.sample_rate
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return sin(phase)
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end
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#### Gain ####
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export Gain
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type Gain <: AudioNode
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active::Bool
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deactivate_cond::Condition
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in_node::AudioNode
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gain::Float32
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function Gain(in_node::AudioNode, gain::Real)
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new(false, Condition(), in_node, gain)
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end
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end
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function render(node::Gain, device_input::AudioBuf, info::DeviceInfo)
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input, child_active = render(node.in_node, device_input, info)
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# TODO: should we check the active flag of the input?
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return input .* node.gain, is_active(node)
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end
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#### AudioMixer ####
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# Mixes a set of inputs equally
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# a convenience alias used in the array of mix inputs
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typealias MaybeAudioNode Union(AudioNode, Nothing)
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const MAX_MIXER_INPUTS = 32
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type MixRenderer <: AudioRenderer
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inputs::Vector{AudioNode}
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end
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typealias AudioMixer AudioNode{MixRenderer}
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AudioMixer{T<:AudioNode}(inputs::Vector{T}) = AudioMixer(MixRenderer(inputs))
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AudioMixer() = AudioMixer(AudioNode[])
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export AudioMixer
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type AudioMixer <: AudioNode
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active::Bool
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deactivate_cond::Condition
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mix_inputs::Array{MaybeAudioNode}
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function AudioMixer{T <: AudioNode}(mix_inputs::Array{T})
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input_array = Array(MaybeAudioNode, MAX_MIXER_INPUTS)
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fill!(input_array, nothing)
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for (i, node) in enumerate(mix_inputs)
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input_array[i] = node
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end
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new(false, Condition(), input_array)
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end
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function AudioMixer()
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AudioMixer(AudioNode[])
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end
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end
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# TODO: at some point we need to figure out what the general API is for wiring
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# up AudioNodes to each other
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function add_input(mixer::AudioMixer, in_node::AudioNode)
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for (i, node) in enumerate(mixer.mix_inputs)
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if node === nothing
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mixer.mix_inputs[i] = in_node
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return
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end
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end
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error("Mixer input array is full")
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end
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# removes the given node from the mix inputs. If the node isn't an input the
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# function returns without error
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function remove_input(mixer::AudioMixer, in_node::AudioNode)
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for (i, node) in enumerate(mixer.mix_inputs)
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if node === in_node
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mixer.mix_inputs[i] = nothing
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return
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end
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end
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# not an error if we didn't find it
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end
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function render(node::AudioMixer, device_input::AudioBuf, info::DeviceInfo)
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function render(node::MixRenderer, device_input::AudioBuf, info::DeviceInfo)
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# TODO: we probably want to pre-allocate this buffer and share between
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# render calls. Unfortunately we don't know the right size when the object
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# is created, so maybe we check the size on every render call and only
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# re-allocate when the size changes? I suppose that's got to be cheaper
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# than the GC and allocation every frame
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mix_buffer = zeros(AudioSample, info.buf_size)
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for in_node in node.mix_inputs
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if in_node !== nothing
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in_buffer, active = render(in_node, device_input, info)
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mix_buffer += in_buffer
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if !active
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remove_input(node, in_node)
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end
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n_inputs = length(node.inputs)
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i = 1
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max_samples = 0
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while i <= n_inputs
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rendered = render(node.inputs[i], device_input, info)::AudioBuf
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nsamples = length(rendered)
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max_samples = max(max_samples, nsamples)
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mix_buffer[1:nsamples] .+= rendered
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if nsamples < info.buf_size
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deleteat!(node.inputs, i)
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n_inputs -= 1
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else
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i += 1
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end
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end
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return mix_buffer, is_active(node)
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return mix_buffer[1:max_samples]
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end
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Base.push!(mixer::AudioMixer, node::AudioNode) = push!(mixer.renderer.inputs, node)
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#### Gain ####
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type GainRenderer <: AudioRenderer
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in_node::AudioNode
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gain::Float32
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end
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function render(node::GainRenderer, device_input::AudioBuf, info::DeviceInfo)
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input = render(node.in_node, device_input, info)
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return input .* node.gain
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end
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typealias Gain AudioNode{GainRenderer}
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Gain(in_node::AudioNode, gain::Real) = Gain(GainRenderer(in_node, gain))
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export Gain
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#### Array Player ####
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# Plays a AudioBuf by rendering it out piece-by-piece
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export ArrayPlayer
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type ArrayPlayer <: AudioNode
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active::Bool
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deactivate_cond::Condition
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type ArrayRenderer <: AudioRenderer
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arr::AudioBuf
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arr_index::Int
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function ArrayPlayer(arr::AudioBuf)
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new(false, Condition(), arr, 1)
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end
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ArrayRenderer(arr::AudioBuf) = new(arr, 1)
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end
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function render(node::ArrayPlayer, device_input::AudioBuf, info::DeviceInfo)
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typealias ArrayPlayer AudioNode{ArrayRenderer}
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ArrayPlayer(arr::AudioBuf) = ArrayPlayer(ArrayRenderer(arr))
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export ArrayPlayer
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function render(node::ArrayRenderer, device_input::AudioBuf, info::DeviceInfo)
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# TODO: this should remove itself from the render tree when playback is
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# complete
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i = node.arr_index
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range_end = min(i + info.buf_size-1, length(node.arr))
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output = node.arr[i:range_end]
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if length(output) < info.buf_size
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# we're finished with the array, pad with zeros and deactivate
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output = vcat(output, zeros(AudioSample, info.buf_size - length(output)))
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deactivate(node)
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end
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node.arr_index = range_end + 1
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return output, is_active(node)
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return output
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end
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# Allow users to play a raw array by wrapping it in an ArrayPlayer
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@ -171,22 +145,22 @@ function play{T <: Unsigned}(arr::Array{T}, args...)
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play(arr, args...)
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end
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#### AudioInput ####
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# Renders incoming audio input from the hardware
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export AudioInput
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type AudioInput <: AudioNode
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active::Bool
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deactivate_cond::Condition
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channel::Int
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function AudioInput(channel::Int)
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new(false, Condition(), channel)
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end
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end
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function render(node::AudioInput, device_input::AudioBuf, info::DeviceInfo)
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@assert size(device_input, 1) == info.buf_size
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return device_input[:, node.channel], is_active(node)
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end
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##### AudioInput ####
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#
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## Renders incoming audio input from the hardware
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#
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#export AudioInput
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#type AudioInput <: AudioNode
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# active::Bool
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# deactivate_cond::Condition
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# channel::Int
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#
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# function AudioInput(channel::Int)
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# new(false, Condition(), channel)
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# end
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#end
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#
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#function render(node::AudioInput, device_input::AudioBuf, info::DeviceInfo)
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# @assert size(device_input, 1) == info.buf_size
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# return device_input[:, node.channel], is_active(node)
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#end
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@ -1,2 +1,13 @@
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*(node::AudioNode, coef::Real) = Gain(node, coef)
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*(coef::Real, node::AudioNode) = Gain(node, coef)
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# multiplying by silence gives silence
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*(in1::NullNode, in2::NullNode) = in1
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*(in1::AudioNode, in2::NullNode) = in2
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*(in1::NullNode, in2::AudioNode) = in1
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+(in1::AudioNode, in2::AudioNode) = AudioMixer([in1, in2])
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# adding silence has no effect
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+(in1::NullNode, in2::NullNode) = in1
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+(in1::AudioNode, in2::NullNode) = in1
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+(in1::NullNode, in2::AudioNode) = in2
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@ -35,13 +35,13 @@ portaudio_inited = false
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################## Types ####################
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type PortAudioStream <: AudioStream
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mixer::AudioMixer
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root::AudioMixer
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info::DeviceInfo
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function PortAudioStream(sample_rate::Int=44100, buf_size::Int=1024)
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init_portaudio()
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mixer = AudioMixer()
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stream = new(mixer, DeviceInfo(sample_rate, buf_size))
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root = AudioMixer()
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stream = new(root, DeviceInfo(sample_rate, buf_size))
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# we need to start up the stream with the portaudio library
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open_portaudio_stream(stream)
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return stream
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@ -91,8 +91,14 @@ function portaudio_task(jl_filedesc::Integer, stream::PortAudioStream)
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jl_rawfd = RawFD(jl_filedesc)
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try
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while true
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# assume the root mixer is always active
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buffer::AudioBuf, _::Bool = render(stream.mixer, buffer, stream.info)
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# assume the root is always active
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rendered = render(stream.root, buffer, stream.info)
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for i in 1:length(rendered)
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buffer[i] = rendered[i]
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end
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for i in (length(rendered)+1):length(buffer)
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buffer[i] = 0.0
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end
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# wake the C code so it knows we've given it some more data
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synchronize_buffer(buffer)
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@ -88,7 +88,6 @@ end
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# through an arbitrary render chain and returns the result as a vector
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function Base.read(file::AudioFile, nframes::Integer = file.sfinfo.frames,
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dtype::Type = Int16)
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arr = []
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@assert file.sfinfo.channels <= 2
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if file.sfinfo.channels == 2
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arr = zeros(dtype, 2, nframes)
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@ -114,11 +113,7 @@ function Base.read(file::AudioFile, nframes::Integer = file.sfinfo.frames,
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file.filePtr, arr, nframes)
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end
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if nread == 0
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return Nothing
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end
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return arr
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return arr[1:nread]
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end
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function Base.write{T}(file::AudioFile, frames::Array{T})
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@ -144,38 +139,34 @@ function Base.write{T}(file::AudioFile, frames::Array{T})
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end
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end
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type FilePlayer <: AudioNode
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active::Bool
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deactivate_cond::Condition
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type FileRenderer <: AudioRenderer
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file::AudioFile
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function FilePlayer(file::AudioFile)
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node = new(false, Condition(), file)
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function FileRenderer(file::AudioFile)
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finalizer(node, node -> close(node.file))
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return node
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end
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function FilePlayer(path::String)
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return FilePlayer(af_open(path))
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end
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end
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function render(node::FilePlayer, device_input::AudioBuf, info::DeviceInfo)
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typealias FilePlayer AudioNode{FileRenderer}
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FilePlayer(file::AudioFile) = FilePlayer(FileRenderer(file))
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FilePlayer(path::String) = FilePlayer(af_open(path))
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function render(node::FileRenderer, device_input::AudioBuf, info::DeviceInfo)
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@assert node.file.sfinfo.samplerate == info.sample_rate
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audio = read(node.file, info.buf_size, AudioSample)
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if audio == Nothing
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deactivate(node)
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return zeros(AudioSample, info.buf_size), is_active(node)
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return AudioSample[]
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end
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# if the file is stereo, mix the two channels together
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if node.file.sfinfo.channels == 2
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return (audio[1, :] / 2) + (audio[2, :] / 2), is_active(node)
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return (audio[1, :] / 2) + (audio[2, :] / 2)
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end
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return audio, is_active(node)
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return audio
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end
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function play(filename::String, args...)
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@ -5,20 +5,22 @@ const TEST_SAMPLERATE = 44100
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const TEST_BUF_SIZE = 1024
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type TestAudioStream <: AudioIO.AudioStream
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mixer::AudioMixer
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root::AudioIO.AudioMixer
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info::AudioIO.DeviceInfo
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function TestAudioStream()
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mixer = AudioMixer()
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new(mixer, AudioIO.DeviceInfo(TEST_SAMPLERATE, TEST_BUF_SIZE))
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root = AudioMixer()
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new(root, AudioIO.DeviceInfo(TEST_SAMPLERATE, TEST_BUF_SIZE))
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end
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end
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# render the stream and return the next block of audio. This is used in testing
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# to simulate the audio callback that's normally called by the device.
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function process(stream::TestAudioStream)
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out_array = zeros(AudioIO.AudioSample, stream.info.buf_size)
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in_array = zeros(AudioIO.AudioSample, stream.info.buf_size)
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out_array, _ = AudioIO.render(stream.mixer, in_array, stream.info)
|
||||
rendered = AudioIO.render(stream.root, in_array, stream.info)
|
||||
out_array[1:length(rendered)] = rendered
|
||||
return out_array
|
||||
end
|
||||
|
||||
|
|
@ -59,18 +61,12 @@ player = play(ui8, test_stream)
|
|||
@test_approx_eq(process(test_stream)[1:255],
|
||||
convert(AudioIO.AudioBuf, linspace(-1.0, 1.0, 255)))
|
||||
|
||||
|
||||
info("Testing AudioNode Stopping...")
|
||||
test_stream = TestAudioStream()
|
||||
node = SinOsc(440)
|
||||
@test !node.active
|
||||
play(node, test_stream)
|
||||
@test node.active
|
||||
process(test_stream)
|
||||
stop(node)
|
||||
@test !node.active
|
||||
# give the render task a chance to clean up
|
||||
process(test_stream)
|
||||
@test process(test_stream) == zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
|
||||
|
||||
info("Testing wav file write/read")
|
||||
|
|
|
|||
|
|
@ -1,23 +1,25 @@
|
|||
using Base.Test
|
||||
using AudioIO
|
||||
import AudioIO.AudioSample
|
||||
import AudioIO.AudioBuf
|
||||
import AudioIO.AudioRenderer
|
||||
import AudioIO.AudioNode
|
||||
import AudioIO.DeviceInfo
|
||||
import AudioIO.render
|
||||
|
||||
test_info = AudioIO.DeviceInfo(44100, 512)
|
||||
dev_input = zeros(AudioIO.AudioSample, test_info.buf_size)
|
||||
test_info = DeviceInfo(44100, 512)
|
||||
dev_input = zeros(AudioSample, test_info.buf_size)
|
||||
|
||||
# A TestNode just renders out 1:buf_size each frame
|
||||
type TestNode <: AudioIO.AudioNode
|
||||
active::Bool
|
||||
deactivate_cond::Condition
|
||||
type TestRenderer <: AudioRenderer end
|
||||
|
||||
function TestNode()
|
||||
return new(false, Condition())
|
||||
end
|
||||
end
|
||||
typealias TestNode AudioNode{TestRenderer}
|
||||
TestNode() = TestNode(TestRenderer())
|
||||
|
||||
function AudioIO.render(node::TestNode,
|
||||
device_input::AudioIO.AudioBuf,
|
||||
info::AudioIO.DeviceInfo)
|
||||
return AudioIO.AudioSample[1:info.buf_size], node.active
|
||||
function render(node::TestRenderer,
|
||||
device_input::AudioBuf,
|
||||
info::DeviceInfo)
|
||||
return AudioSample[1:info.buf_size]
|
||||
end
|
||||
|
||||
#### AudioMixer Tests ####
|
||||
|
|
@ -27,64 +29,64 @@ end
|
|||
|
||||
info("Testing AudioMixer...")
|
||||
mix = AudioMixer()
|
||||
render_output, active = AudioIO.render(mix, dev_input, test_info)
|
||||
@test render_output == zeros(AudioIO.AudioSample, test_info.buf_size)
|
||||
render_output = render(mix, dev_input, test_info)
|
||||
@test render_output == AudioSample[]
|
||||
|
||||
testnode = TestNode()
|
||||
mix = AudioMixer([testnode])
|
||||
render_output, active = AudioIO.render(mix, dev_input, test_info)
|
||||
@test render_output == AudioIO.AudioSample[1:test_info.buf_size]
|
||||
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, active = AudioIO.render(mix, dev_input, test_info)
|
||||
render_output = render(mix, dev_input, test_info)
|
||||
# make sure the two inputs are being added together
|
||||
@test render_output == 2 * AudioIO.AudioSample[1:test_info.buf_size]
|
||||
@test render_output == 2 * AudioSample[1:test_info.buf_size]
|
||||
|
||||
# now we'll stop one of the inputs and make sure it gets removed
|
||||
# TODO: this test should depend on the render output, not on the internals of
|
||||
# the mixer
|
||||
stop(test1)
|
||||
AudioIO.render(mix, dev_input, test_info)
|
||||
@test !in(test1, mix.mix_inputs)
|
||||
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, active = AudioIO.render(mix, dev_input, test_info)
|
||||
@test !active
|
||||
render_output = render(mix, dev_input, test_info)
|
||||
@test render_output == AudioSample[]
|
||||
|
||||
info("Testing SinOSC...")
|
||||
freq = 440
|
||||
t = linspace(1 / test_info.sample_rate,
|
||||
test_info.buf_size / test_info.sample_rate,
|
||||
t = linspace(0,
|
||||
(test_info.buf_size-1) / test_info.sample_rate,
|
||||
test_info.buf_size)
|
||||
test_vect = convert(AudioIO.AudioBuf, sin(2pi * t * freq))
|
||||
test_vect = convert(AudioBuf, sin(2pi * t * freq))
|
||||
osc = SinOsc(freq)
|
||||
render_output, active = AudioIO.render(osc, dev_input, test_info)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@test_approx_eq(render_output, test_vect)
|
||||
stop(osc)
|
||||
render_output, active = AudioIO.render(osc, dev_input, test_info)
|
||||
@test !active
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@test render_output == AudioSample[]
|
||||
|
||||
info("Testing ArrayPlayer...")
|
||||
v = rand(AudioIO.AudioSample, 44100)
|
||||
v = rand(AudioSample, 44100)
|
||||
player = ArrayPlayer(v)
|
||||
player.active = true
|
||||
render_output, active = AudioIO.render(player, dev_input, test_info)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@test render_output == v[1:test_info.buf_size]
|
||||
@test active
|
||||
render_output, active = AudioIO.render(player, dev_input, test_info)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@test render_output == v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
|
||||
@test active
|
||||
stop(player)
|
||||
render_output, active = AudioIO.render(player, dev_input, test_info)
|
||||
@test !active
|
||||
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(AudioIO.AudioSample, test_info.buf_size + 1)
|
||||
v = rand(AudioSample, test_info.buf_size + 1)
|
||||
player = ArrayPlayer(v)
|
||||
player.active = true
|
||||
_, active = AudioIO.render(player, dev_input, test_info)
|
||||
@test active
|
||||
_, active = AudioIO.render(player, dev_input, test_info)
|
||||
@test !active
|
||||
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]
|
||||
|
|
|
|||
Loading…
Reference in a new issue