more fixes to get the tests running without deprecations on 0.4 and 0.3
This commit is contained in:
Spencer Russell
parent
826fdafe5f
commit
c4dfef9178
8 changed files with 110 additions and 111 deletions
22
.travis.yml
22
.travis.yml
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@ -1,21 +1,21 @@
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language: cpp
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language: julia
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compiler:
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- clang
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notifications:
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email: spencer.f.russell@gmail.com
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env:
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- JULIAVERSION="julianightlies"
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- JULIAVERSION="juliareleases"
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before_install:
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- sudo add-apt-repository ppa:staticfloat/julia-deps -y
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- sudo add-apt-repository ppa:staticfloat/$JULIAVERSION -y
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- sudo apt-get update -qq -y
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- sudo apt-get install libpcre3-dev julia -y
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- if [[ -a .git/shallow ]]; then git fetch --unshallow; fi
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julia:
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- 0.3
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- 0.4
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# - nightly
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# Fails with:
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# LoadError: ccall: could not find function jl_read_sonames
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# while loading /home/travis/.julia/v0.5/AudioIO/deps/build.jl, in expression starting on line 29
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script:
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- if [[ -a .git/shallow ]]; then git fetch --unshallow; fi
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- julia -e 'Pkg.init()'
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- julia -e 'Pkg.add("BinDeps"); Pkg.checkout("BinDeps")' # latest master needed for Pacman support
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- julia -e 'Pkg.clone(pwd()); Pkg.build("AudioIO")'
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- julia -e 'Pkg.test("AudioIO", coverage=true)'
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- julia -e 'Pkg.add("FactCheck")' # add FactCheck manually because we're not using Pkg.test()
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- julia --code-coverage=user test/runtests.jl # Pkg.test disables inlining when enabling coverage, which kills our allocation tests
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after_success:
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- if [ $JULIAVERSION = "juliareleases" ]; then julia -e 'cd(Pkg.dir("AudioIO")); Pkg.add("Coverage"); using Coverage; Coveralls.submit(Coveralls.process_folder())'; fi
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5
deps/build.jl
vendored
5
deps/build.jl
vendored
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@ -1,4 +1,5 @@
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using BinDeps
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using Compat
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@BinDeps.setup
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@ -26,5 +27,5 @@ end
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provides(WinRPM.RPM, "libsndfile1", libsndfile, os = :Windows)
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end
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@BinDeps.install [:libportaudio => :libportaudio,
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:libsndfile => :libsndfile]
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@BinDeps.install @compat(Dict(:libportaudio => :libportaudio,
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:libsndfile => :libsndfile))
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@ -228,7 +228,7 @@ function play(arr::AudioBuf, args...)
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end
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# If the array is the wrong floating type, convert it
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@compat function play{T <: AbstractFloat}(arr::Array{T}, args...)
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function play{T <: AbstractFloat}(arr::Array{T}, args...)
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arr = convert(AudioBuf, arr)
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play(arr, args...)
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end
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@ -306,7 +306,7 @@ export LinRamp
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function render(node::LinRampRenderer, device_input::AudioBuf, info::DeviceInfo)
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# Resize buffer if (1) it's too small or (2) we've hit the end of the ramp
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ramp_samples::Int = int(node.duration * info.sample_rate)
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ramp_samples::Int = round(Int, node.duration * info.sample_rate)
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block_samples = min(ramp_samples, info.buf_size)
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if length(node.buf) != block_samples
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resize!(node.buf, block_samples)
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@ -1,7 +1,7 @@
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typealias PaTime Cdouble
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typealias PaError Cint
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typealias PaSampleFormat Culong
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# PaStream is always used as an opaque type, so we're always dealing
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# PaStream is always used as an opaque type, so we're always dealing
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# with the pointer
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typealias PaStream Ptr{Void}
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typealias PaDeviceIndex Cint
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@ -23,7 +23,7 @@ const paInt8 = convert(PaSampleFormat, 0x10)
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const paUInt8 = convert(PaSampleFormat, 0x20)
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# PaHostApiTypeId values
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@compat const pa_host_api_names = (
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@compat const pa_host_api_names = Dict(
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0 => "In Development", # use while developing support for a new host API
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1 => "Direct Sound",
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2 => "MME",
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@ -95,31 +95,31 @@ end
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The stream is unidirectional, either inout or default output
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see http://portaudio.com/docs/v19-doxydocs/portaudio_8h.html
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"""
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function Pa_OpenStream(device::PaDeviceIndex,
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function Pa_OpenStream(device::PaDeviceIndex,
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channels::Cint, input::Bool,
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sampleFormat::PaSampleFormat,
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sampleRate::Cdouble, framesPerBuffer::Culong)
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streamPtr::Array{PaStream} = PaStream[0]
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ioParameters = Pa_StreamParameters(device, channels,
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sampleFormat, PaTime(0.001),
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ioParameters = Pa_StreamParameters(device, channels,
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sampleFormat, PaTime(0.001),
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Ptr{Void}(0))
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if input
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err = ccall((:Pa_OpenStream, libportaudio), PaError,
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err = ccall((:Pa_OpenStream, libportaudio), PaError,
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(PaStream, Ref{Pa_StreamParameters}, Ptr{Void},
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Cdouble, Culong, Culong,
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Cdouble, Culong, Culong,
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Ptr{PaStreamCallback}, Ptr{Void}),
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streamPtr, ioParameters, Ptr{Void}(0),
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sampleRate, framesPerBuffer, 0,
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sampleRate, framesPerBuffer, 0,
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Ptr{PaStreamCallback}(0), Ptr{Void}(0))
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else
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err = ccall((:Pa_OpenStream, libportaudio), PaError,
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err = ccall((:Pa_OpenStream, libportaudio), PaError,
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(PaStream, Ptr{Void}, Ref{Pa_StreamParameters},
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Cdouble, Culong, Culong,
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Ptr{PaStreamCallback}, Ptr{Void}),
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streamPtr, Ptr{Void}(0), ioParameters,
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sampleRate, framesPerBuffer, 0,
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sampleRate, framesPerBuffer, 0,
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Ptr{PaStreamCallback}(0), Ptr{Void}(0))
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end
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end
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handle_status(err)
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streamPtr[1]
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end
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@ -134,7 +134,7 @@ type Pa_AudioStream <: AudioStream
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sbuffer_output_waiting::Integer
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parent_may_use_buffer::Bool
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"""
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"""
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Get device parameters needed for opening with portaudio
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default is input as 44100/16bit int, same as CD audio type input
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"""
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@ -150,7 +150,7 @@ type Pa_AudioStream <: AudioStream
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root = AudioMixer()
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datatype = PaSampleFormat_to_T(sample_format)
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sbuf = ones(datatype, framesPerBuffer)
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this = new(root, DeviceInfo(sample_rate, framesPerBuffer),
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this = new(root, DeviceInfo(sample_rate, framesPerBuffer),
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show_warnings, stream, sample_format, sbuf, 0, false)
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info("Scheduling PortAudio Render Task...")
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if input
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@ -170,7 +170,7 @@ function read_Pa_AudioStream(stream::Pa_AudioStream)
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while stream.parent_may_use_buffer == false
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sleep(0.001)
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end
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buffer = deepcopy(stream.sbuffer)
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buffer = deepcopy(stream.sbuffer)
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stream.parent_may_use_buffer = false
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return buffer
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end
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@ -234,7 +234,7 @@ function portaudio_task(stream::PortAudioStream)
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end
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"""
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Helper function to make the right type of buffer for various
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Helper function to make the right type of buffer for various
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sample formats. Converts PaSampleFormat to a typeof
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"""
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function PaSampleFormat_to_T(fmt::PaSampleFormat)
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@ -276,7 +276,7 @@ function pa_input_task(stream::Pa_AudioStream)
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end
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err = ccall((:Pa_ReadStream, libportaudio), PaError,
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(PaStream, Ptr{Void}, Culong),
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stream.stream, buffer, n)
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stream.stream, buffer, n)
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handle_status(err, stream.show_warnings)
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stream.sbuffer[1: n] = buffer[1: n]
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stream.parent_may_use_buffer = true
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@ -303,12 +303,12 @@ function pa_output_task(stream::Pa_AudioStream)
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end
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if (navail > 1) & (stream.parent_may_use_buffer == false) &
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(Pa_GetStreamWriteAvailable(stream.stream) < navail)
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Pa_WriteStream(stream.stream, stream.sbuffer,
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Pa_WriteStream(stream.stream, stream.sbuffer,
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navail, stream.show_warnings)
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stream.parent_may_use_buffer = true
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else
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sleep(0.005)
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end
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end
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end
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catch ex
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warn("Audio Output Task died with exception: $ex")
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@ -338,7 +338,7 @@ type PaHostApiInfo
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defaultOutputDevice::PaDeviceIndex
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end
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@compat type PortAudioInterface <: AudioInterface
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type PortAudioInterface <: AudioInterface
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name::AbstractString
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host_api::AbstractString
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max_input_channels::Int
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@ -16,7 +16,7 @@ const SF_SEEK_SET = 0
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const SF_SEEK_CUR = 1
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const SF_SEEK_END = 2
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@compat const EXT_TO_FORMAT = (
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@compat const EXT_TO_FORMAT = Dict(
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".wav" => SF_FORMAT_WAV,
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".flac" => SF_FORMAT_FLAC
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)
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@ -28,12 +28,6 @@ type SF_INFO
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format::Int32
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sections::Int32
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seekable::Int32
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function SF_INFO(frames::Integer, samplerate::Integer, channels::Integer,
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format::Integer, sections::Integer, seekable::Integer)
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new(int64(frames), int32(samplerate), int32(channels), int32(format),
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int32(sections), int32(seekable))
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end
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end
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type AudioFile
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@ -45,7 +39,7 @@ samplerate(f::AudioFile) = f.sfinfo.samplerate
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# AudioIO.open is part of the public API, but is not exported so that it
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# doesn't conflict with Base.open
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@compat function open(path::AbstractString, mode::AbstractString = "r",
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function open(path::AbstractString, mode::AbstractString = "r",
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sampleRate::Integer = 44100, channels::Integer = 1,
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format::Integer = 0)
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@assert channels <= 2
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@ -66,11 +60,11 @@ samplerate(f::AudioFile) = f.sfinfo.samplerate
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end
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filePtr = ccall((:sf_open, libsndfile), Ptr{Void},
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(Ptr{Uint8}, Int32, Ptr{SF_INFO}),
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(Ptr{UInt8}, Int32, Ptr{SF_INFO}),
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path, file_mode, &sfinfo)
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if filePtr == C_NULL
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errmsg = ccall((:sf_strerror, libsndfile), Ptr{Uint8}, (Ptr{Void},), filePtr)
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errmsg = ccall((:sf_strerror, libsndfile), Ptr{UInt8}, (Ptr{Void},), filePtr)
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error(bytestring(errmsg))
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end
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@ -132,7 +126,7 @@ Base.read(file::AudioFile) = Base.read(file, Int16)
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function Base.write{T}(file::AudioFile, frames::Array{T})
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@assert file.sfinfo.channels <= 2
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nframes = int(length(frames) / file.sfinfo.channels)
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nframes = round(Int, length(frames) / file.sfinfo.channels)
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if T == Int16
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return ccall((:sf_writef_short, libsndfile), Int64,
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@ -180,7 +174,7 @@ end
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typealias FilePlayer AudioNode{FileRenderer}
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FilePlayer(file::AudioFile) = FilePlayer(FileRenderer(file))
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@compat FilePlayer(path::AbstractString) = FilePlayer(AudioIO.open(path))
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FilePlayer(path::AbstractString) = FilePlayer(AudioIO.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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@ -204,7 +198,7 @@ function render(node::FileRenderer, device_input::AudioBuf, info::DeviceInfo)
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end
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end
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@compat function play(filename::AbstractString, args...)
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function play(filename::AbstractString, args...)
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player = FilePlayer(filename)
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play(player, args...)
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end
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@ -1,6 +1,7 @@
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module TestAudioIO
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using FactCheck
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using Compat
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using AudioIO
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import AudioIO.AudioBuf
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@ -43,32 +44,32 @@ facts("Array playback") do
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f32 = convert(Array{Float32}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f32, test_stream)
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@fact process(test_stream) => f32[1:TEST_BUF_SIZE]
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@fact process(test_stream) --> f32[1:TEST_BUF_SIZE]
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end
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context("Playing Float64 arrays") do
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f64 = convert(Array{Float64}, sin(phase))
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test_stream = TestAudioStream()
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player = play(f64, test_stream)
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@fact process(test_stream) => convert(AudioBuf, f64[1:TEST_BUF_SIZE])
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@fact process(test_stream) --> convert(AudioBuf, f64[1:TEST_BUF_SIZE])
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end
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context("Playing Int8(Signed) arrays") do
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i8 = Int8[-127:127]
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i8 = Int8[-127:127;]
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test_stream = TestAudioStream()
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player = play(i8, test_stream)
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@fact process(test_stream)[1:255] =>
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mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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@fact process(test_stream)[1:255] -->
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mse(convert(AudioBuf, collect(linspace(-1.0, 1.0, 255))))
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end
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context("Playing Uint8(Unsigned) arrays") do
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context("Playing UInt8(Unsigned) arrays") do
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# for unsigned 8-bit audio silence is represented as 128, so the symmetric range
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# is 1-255
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ui8 = Uint8[1:255]
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ui8 = UInt8[1:255;]
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test_stream = TestAudioStream()
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player = play(ui8, test_stream)
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@fact process(test_stream)[1:255] =>
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mse(convert(AudioBuf, linspace(-1.0, 1.0, 255)))
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@fact process(test_stream)[1:255] -->
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mse(convert(AudioBuf, collect(linspace(-1.0, 1.0, 255))))
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end
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end
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@ -78,12 +79,12 @@ facts("AudioNode Stopping") do
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play(node, test_stream)
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process(test_stream)
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stop(node)
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@fact process(test_stream) => zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
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@fact process(test_stream) --> zeros(AudioIO.AudioSample, TEST_BUF_SIZE)
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end
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facts("Audio Device Listing") do
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# there aren't any devices on the Travis machine so just test that this doesn't crash
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@fact get_audio_devices() => issubtype(Array)
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@fact get_audio_devices() --> issubtype(Array)
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end
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end # module TestAudioIO
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@ -1,5 +1,6 @@
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module TestAudioIONodes
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using Compat
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using FactCheck
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using AudioIO
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import AudioIO: AudioSample, AudioBuf, AudioRenderer, AudioNode
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@ -10,7 +11,7 @@ include("testhelpers.jl")
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# A TestNode just renders out 1:buf_size each frame
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type TestRenderer <: AudioRenderer
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buf::AudioBuf
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TestRenderer(buf_size::Integer) = new(AudioSample[1:buf_size])
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TestRenderer(buf_size::Integer) = new(AudioSample[1:buf_size;])
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end
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typealias TestNode AudioNode{TestRenderer}
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@ -31,7 +32,7 @@ facts("Validating TestNode allocation") do
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test = TestNode(test_info.buf_size)
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# JIT
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render(test, dev_input, test_info)
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@fact (@allocated render(test, dev_input, test_info)) => 16
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@fact (@allocated render(test, dev_input, test_info)) --> 16
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end
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#### AudioMixer Tests ####
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@ -43,16 +44,17 @@ facts("AudioMixer") do
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context("0 Input Mixer") do
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mix = AudioMixer()
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[]
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@fact (@allocated render(mix, dev_input, test_info)) => 48
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@fact render_output --> AudioSample[]
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# 0.4 uses 64 bytes, 0.3 uses 48
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@fact (@allocated render(mix, dev_input, test_info)) --> less_than(65)
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end
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context("1 Input Mixer") do
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testnode = TestNode(test_info.buf_size)
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mix = AudioMixer([testnode])
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[1:test_info.buf_size]
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@fact (@allocated render(mix, dev_input, test_info)) => 64
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@fact render_output --> AudioSample[1:test_info.buf_size;]
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@fact (@allocated render(mix, dev_input, test_info)) --> 64
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end
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context("2 Input Mixer") do
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@ -61,17 +63,17 @@ facts("AudioMixer") do
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mix = AudioMixer([test1, test2])
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@fact render_output => 2 * AudioSample[1:test_info.buf_size]
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@fact (@allocated render(mix, dev_input, test_info)) => 96
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@fact render_output --> 2 * AudioSample[1:test_info.buf_size;]
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@fact (@allocated render(mix, dev_input, test_info)) --> 96
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# now we'll stop one of the inputs and make sure it gets removed
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stop(test1)
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render_output = render(mix, dev_input, test_info)
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# make sure the two inputs are being added together
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@fact render_output => AudioSample[1:test_info.buf_size]
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@fact render_output --> AudioSample[1:test_info.buf_size;]
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stop(mix)
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render_output = render(mix, dev_input, test_info)
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@fact render_output => AudioSample[]
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@fact render_output --> AudioSample[]
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end
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end
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@ -81,25 +83,25 @@ facts("SinOSC") do
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freq = 440
|
||||
# note that this range includes the end, which is why there are
|
||||
# sample_rate+1 samples
|
||||
t = linspace(0, 1, int(test_info.sample_rate+1))
|
||||
t = linspace(0, 1, round(Int, test_info.sample_rate+1))
|
||||
test_vect = convert(AudioBuf, sin(2pi * t * freq))
|
||||
context("Fixed Frequency") do
|
||||
osc = SinOsc(freq)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@fact mse(render_output, test_vect[1:test_info.buf_size]) =>
|
||||
@fact mse(render_output, test_vect[1:test_info.buf_size]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@fact mse(render_output,
|
||||
test_vect[test_info.buf_size+1:2*test_info.buf_size]) =>
|
||||
test_vect[test_info.buf_size+1:2*test_info.buf_size]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
@fact (@allocated render(osc, dev_input, test_info)) => 64
|
||||
@fact (@allocated render(osc, dev_input, test_info)) --> 64
|
||||
stop(osc)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@fact render_output => AudioSample[]
|
||||
@fact render_output --> AudioSample[]
|
||||
end
|
||||
|
||||
context("Testing SinOsc with signal input") do
|
||||
t = linspace(0, 1, int(test_info.sample_rate+1))
|
||||
t = linspace(0, 1, round(Int, 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
|
||||
|
|
@ -112,14 +114,14 @@ facts("SinOSC") do
|
|||
freq = LinRamp(440, 110, 1)
|
||||
osc = SinOsc(freq)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@fact mse(render_output, expected[1:test_info.buf_size]) =>
|
||||
@fact mse(render_output, expected[1:test_info.buf_size]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
render_output = render(osc, dev_input, test_info)
|
||||
@fact mse(render_output,
|
||||
expected[test_info.buf_size+1:2*test_info.buf_size]) =>
|
||||
expected[test_info.buf_size+1:2*test_info.buf_size]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
# give a bigger budget here because we're rendering 2 nodes
|
||||
@fact (@allocated render(osc, dev_input, test_info)) => 160
|
||||
@fact (@allocated render(osc, dev_input, test_info)) --> 160
|
||||
end
|
||||
end
|
||||
|
||||
|
|
@ -127,7 +129,7 @@ facts("AudioInput") do
|
|||
node = AudioInput()
|
||||
test_data = rand(AudioSample, test_info.buf_size)
|
||||
render_output = render(node, test_data, test_info)
|
||||
@fact render_output => test_data
|
||||
@fact render_output --> test_data
|
||||
end
|
||||
|
||||
facts("ArrayPlayer") do
|
||||
|
|
@ -135,13 +137,13 @@ facts("ArrayPlayer") do
|
|||
v = rand(AudioSample, 44100)
|
||||
player = ArrayPlayer(v)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@fact render_output => v[1:test_info.buf_size]
|
||||
@fact render_output --> v[1:test_info.buf_size]
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@fact render_output => v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
|
||||
@fact (@allocated render(player, dev_input, test_info)) => 192
|
||||
@fact render_output --> v[(test_info.buf_size + 1) : (2*test_info.buf_size)]
|
||||
@fact (@allocated render(player, dev_input, test_info)) --> 192
|
||||
stop(player)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@fact render_output => AudioSample[]
|
||||
@fact render_output --> AudioSample[]
|
||||
end
|
||||
|
||||
context("testing end of vector") do
|
||||
|
|
@ -150,7 +152,7 @@ facts("ArrayPlayer") do
|
|||
player = ArrayPlayer(v)
|
||||
render(player, dev_input, test_info)
|
||||
render_output = render(player, dev_input, test_info)
|
||||
@fact render_output => v[test_info.buf_size+1:end]
|
||||
@fact render_output --> v[test_info.buf_size+1:end]
|
||||
end
|
||||
end
|
||||
|
||||
|
|
@ -158,35 +160,35 @@ facts("Gain") do
|
|||
context("Constant Gain") do
|
||||
gained = TestNode(test_info.buf_size) * 0.75
|
||||
render_output = render(gained, dev_input, test_info)
|
||||
@fact render_output => 0.75 * AudioSample[1:test_info.buf_size]
|
||||
@fact (@allocated render(gained, dev_input, test_info)) => 32
|
||||
@fact render_output --> 0.75 * AudioSample[1:test_info.buf_size;]
|
||||
@fact (@allocated render(gained, dev_input, test_info)) --> 32
|
||||
end
|
||||
context("Gain by a Signal") do
|
||||
gained = TestNode(test_info.buf_size) * TestNode(test_info.buf_size)
|
||||
render_output = render(gained, dev_input, test_info)
|
||||
@fact render_output => AudioSample[1:test_info.buf_size] .* AudioSample[1:test_info.buf_size]
|
||||
@fact (@allocated render(gained, dev_input, test_info)) => 48
|
||||
@fact render_output --> AudioSample[1:test_info.buf_size;] .* AudioSample[1:test_info.buf_size;]
|
||||
@fact (@allocated render(gained, dev_input, test_info)) --> 48
|
||||
end
|
||||
end
|
||||
|
||||
facts("LinRamp") do
|
||||
ramp = LinRamp(0.25, 0.80, 1)
|
||||
expected = convert(AudioBuf, linspace(0.25, 0.80, int(test_info.sample_rate+1)))
|
||||
expected = convert(AudioBuf, collect(linspace(0.25, 0.80, round(Int, test_info.sample_rate+1))))
|
||||
render_output = render(ramp, dev_input, test_info)
|
||||
@fact mse(render_output, expected[1:test_info.buf_size]) =>
|
||||
@fact mse(render_output, expected[1:test_info.buf_size]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
render_output = render(ramp, dev_input, test_info)
|
||||
@fact mse(render_output,
|
||||
expected[(test_info.buf_size+1):(2*test_info.buf_size)]) =>
|
||||
expected[(test_info.buf_size+1):(2*test_info.buf_size)]) -->
|
||||
lessthan(MSE_THRESH)
|
||||
@fact (@allocated render(ramp, dev_input, test_info)) => 64
|
||||
@fact (@allocated render(ramp, dev_input, test_info)) --> 64
|
||||
end
|
||||
|
||||
facts("Offset") do
|
||||
offs = TestNode(test_info.buf_size) + 0.5
|
||||
render_output = render(offs, dev_input, test_info)
|
||||
@fact render_output => 0.5 + AudioSample[1:test_info.buf_size]
|
||||
@fact (@allocated render(offs, dev_input, test_info)) => 32
|
||||
@fact render_output --> 0.5 + AudioSample[1:test_info.buf_size;]
|
||||
@fact (@allocated render(offs, dev_input, test_info)) --> 32
|
||||
end
|
||||
|
||||
end # module TestAudioIONodes
|
||||
|
|
|
|||
|
|
@ -1,19 +1,20 @@
|
|||
module TestSndfile
|
||||
|
||||
include("testhelpers.jl")
|
||||
|
||||
using AudioIO
|
||||
using Compat
|
||||
using FactCheck
|
||||
using AudioIO
|
||||
import AudioIO: DeviceInfo, render, AudioSample, AudioBuf
|
||||
|
||||
include("testhelpers.jl")
|
||||
|
||||
facts("WAV file write/read") do
|
||||
fname = Pkg.dir("AudioIO", "test", "sinwave.wav")
|
||||
|
||||
srate = 44100
|
||||
freq = 440
|
||||
t = [0 : 2 * srate - 1] / srate
|
||||
t = collect(0 : 2 * srate - 1) / srate
|
||||
phase = 2 * pi * freq * t
|
||||
reference = int16((2 ^ 15 - 1) * sin(phase))
|
||||
reference = round(Int16, (2 ^ 15 - 1) * sin(phase))
|
||||
|
||||
AudioIO.open(fname, "w") do f
|
||||
write(f, reference)
|
||||
|
|
@ -21,12 +22,12 @@ facts("WAV file write/read") do
|
|||
|
||||
# test basic reading
|
||||
AudioIO.open(fname) do f
|
||||
@fact f.sfinfo.channels => 1
|
||||
@fact f.sfinfo.frames => 2 * srate
|
||||
@fact f.sfinfo.channels --> 1
|
||||
@fact f.sfinfo.frames --> 2 * srate
|
||||
actual = read(f)
|
||||
@fact length(reference) => length(actual)
|
||||
@fact reference => actual[:, 1]
|
||||
@fact samplerate(f) => srate
|
||||
@fact length(reference) --> length(actual)
|
||||
@fact reference --> actual[:, 1]
|
||||
@fact samplerate(f) --> srate
|
||||
end
|
||||
|
||||
# test seeking
|
||||
|
|
@ -41,21 +42,21 @@ facts("WAV file write/read") do
|
|||
# convert to floating point because that's what AudioIO uses natively
|
||||
expected = convert(AudioBuf, reference ./ (2^15))
|
||||
buf = render(node, input, test_info)
|
||||
@fact expected[1:bufsize] => buf[1:bufsize]
|
||||
@fact expected[1:bufsize] --> buf[1:bufsize]
|
||||
buf = render(node, input, test_info)
|
||||
@fact expected[bufsize+1:2*bufsize] => buf[1:bufsize]
|
||||
@fact expected[bufsize+1:2*bufsize] --> buf[1:bufsize]
|
||||
end
|
||||
end
|
||||
|
||||
facts("Stereo file reading") do
|
||||
fname = Pkg.dir("AudioIO", "test", "440left_880right.wav")
|
||||
srate = 44100
|
||||
t = [0 : 2 * srate - 1] / srate
|
||||
expected = int16((2^15-1) * hcat(sin(2pi*t*440), sin(2pi*t*880)))
|
||||
t = collect(0:(2 * srate - 1)) / srate
|
||||
expected = round(Int16, (2^15-1) * hcat(sin(2pi*t*440), sin(2pi*t*880)))
|
||||
|
||||
AudioIO.open(fname) do f
|
||||
buf = read(f)
|
||||
@fact buf => mse(expected, 5)
|
||||
@fact buf --> mse(expected, 5)
|
||||
end
|
||||
end
|
||||
|
||||
|
|
@ -67,15 +68,15 @@ facts("Stereo file rendering") do
|
|||
bufsize = 1024
|
||||
input = zeros(AudioSample, bufsize)
|
||||
test_info = DeviceInfo(srate, bufsize)
|
||||
t = [0 : 2 * srate - 1] / srate
|
||||
t = collect(0 : 2 * srate - 1) / srate
|
||||
expected = convert(AudioBuf, 0.5 * (sin(2pi*t*440) + sin(2pi*t*880)))
|
||||
|
||||
AudioIO.open(fname) do f
|
||||
node = FilePlayer(f)
|
||||
buf = render(node, input, test_info)
|
||||
@fact buf[1:bufsize] => mse(expected[1:bufsize])
|
||||
@fact buf[1:bufsize] --> mse(expected[1:bufsize])
|
||||
buf = render(node, input, test_info)
|
||||
@fact buf[1:bufsize] => mse(expected[bufsize+1:2*bufsize])
|
||||
@fact buf[1:bufsize] --> mse(expected[bufsize+1:2*bufsize])
|
||||
end
|
||||
end
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue