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Merge pull request #72 from spotify/psobot/gsm-compressor
Add GSMFullRateCompressor plugin.
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@@ -16,6 +16,8 @@ | |
| */ | ||
| #pragma once | ||
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| #pragma once | ||
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| #include "../JucePlugin.h" | ||
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| namespace Pedalboard { | ||
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| /* | ||
| * pedalboard | ||
| * Copyright 2022 Spotify AB | ||
| * | ||
| * Licensed under the GNU Public License, Version 3.0 (the "License"); | ||
| * you may not use this file except in compliance with the License. | ||
| * You may obtain a copy of the License at | ||
| * | ||
| * https://www.gnu.org/licenses/gpl-3.0.html | ||
| * | ||
| * Unless required by applicable law or agreed to in writing, software | ||
| * distributed under the License is distributed on an "AS IS" BASIS, | ||
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| * See the License for the specific language governing permissions and | ||
| * limitations under the License. | ||
| */ | ||
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| #include "../Plugin.h" | ||
| #include "../plugin_templates/FixedBlockSize.h" | ||
| #include "../plugin_templates/ForceMono.h" | ||
| #include "../plugin_templates/PrimeWithSilence.h" | ||
| #include "../plugin_templates/Resample.h" | ||
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| extern "C" { | ||
| #include <gsm.h> | ||
| } | ||
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| namespace Pedalboard { | ||
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| /* | ||
| * A small C++ wrapper around the C-based libgsm object. | ||
| * Used mostly to avoid leaking memory. | ||
| */ | ||
| class GSMWrapper { | ||
| public: | ||
| GSMWrapper() {} | ||
| ~GSMWrapper() { reset(); } | ||
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| operator bool() const { return _gsm != nullptr; } | ||
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| void reset() { | ||
| gsm_destroy(_gsm); | ||
| _gsm = nullptr; | ||
| } | ||
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| gsm getContext() { | ||
| if (!_gsm) | ||
| _gsm = gsm_create(); | ||
| return _gsm; | ||
| } | ||
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| private: | ||
| gsm _gsm = nullptr; | ||
| }; | ||
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| class GSMFullRateCompressorInternal : public Plugin { | ||
| public: | ||
| virtual ~GSMFullRateCompressorInternal(){}; | ||
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| virtual void prepare(const juce::dsp::ProcessSpec &spec) override { | ||
| bool specChanged = lastSpec.sampleRate != spec.sampleRate || | ||
| lastSpec.maximumBlockSize < spec.maximumBlockSize || | ||
| lastSpec.numChannels != spec.numChannels; | ||
| if (!encoder || specChanged) { | ||
| reset(); | ||
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| if (spec.sampleRate != GSM_SAMPLE_RATE) { | ||
| throw std::runtime_error("GSMCompressor plugin must be run at " + | ||
| std::to_string(GSM_SAMPLE_RATE) + "Hz!"); | ||
| } | ||
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| if (!encoder.getContext()) { | ||
| throw std::runtime_error("Failed to initialize GSM encoder."); | ||
| } | ||
| if (!decoder.getContext()) { | ||
| throw std::runtime_error("Failed to initialize GSM decoder."); | ||
| } | ||
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| lastSpec = spec; | ||
| } | ||
| } | ||
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| int process( | ||
| const juce::dsp::ProcessContextReplacing<float> &context) override final { | ||
| auto ioBlock = context.getOutputBlock(); | ||
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| if (ioBlock.getNumSamples() != GSM_FRAME_SIZE_SAMPLES) { | ||
| throw std::runtime_error("GSMCompressor plugin must be passed exactly " + | ||
| std::to_string(GSM_FRAME_SIZE_SAMPLES) + | ||
| " at a time."); | ||
| } | ||
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| if (ioBlock.getNumChannels() != 1) { | ||
| throw std::runtime_error( | ||
| "GSMCompressor plugin must be passed mono input!"); | ||
| } | ||
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| // Convert samples to signed 16-bit integer first, | ||
| // then pass to the GSM Encoder, then immediately back | ||
| // around to the GSM decoder. | ||
| short frame[GSM_FRAME_SIZE_SAMPLES]; | ||
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| juce::AudioDataConverters::convertFloatToInt16LE( | ||
| ioBlock.getChannelPointer(0), frame, GSM_FRAME_SIZE_SAMPLES); | ||
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| // Actually do the GSM processing! | ||
| gsm_frame encodedFrame; | ||
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| gsm_encode(encoder.getContext(), frame, encodedFrame); | ||
| if (gsm_decode(decoder.getContext(), encodedFrame, frame) < 0) { | ||
| throw std::runtime_error("GSM decoder could not decode frame!"); | ||
| } | ||
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| juce::AudioDataConverters::convertInt16LEToFloat( | ||
| frame, ioBlock.getChannelPointer(0), GSM_FRAME_SIZE_SAMPLES); | ||
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| return GSM_FRAME_SIZE_SAMPLES; | ||
| } | ||
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| void reset() override final { | ||
| encoder.reset(); | ||
| decoder.reset(); | ||
| } | ||
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| static constexpr size_t GSM_FRAME_SIZE_SAMPLES = 160; | ||
| static constexpr int GSM_SAMPLE_RATE = 8000; | ||
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| private: | ||
| GSMWrapper encoder; | ||
| GSMWrapper decoder; | ||
| }; | ||
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| /** | ||
| * Use the GSMFullRateCompressorInternal plugin, but: | ||
| * - ensure that it only ever sees fixed-size blocks of 160 samples | ||
| * - prime the input with a single block of silence | ||
| * - resample whatever input sample rate is provided down to 8kHz | ||
| * - only provide mono input to the plugin, and copy the mono signal | ||
| * back to stereo if necessary | ||
| */ | ||
| using GSMFullRateCompressor = ForceMono<Resample< | ||
| PrimeWithSilence< | ||
| FixedBlockSize<GSMFullRateCompressorInternal, | ||
| GSMFullRateCompressorInternal::GSM_FRAME_SIZE_SAMPLES>, | ||
| float, GSMFullRateCompressorInternal::GSM_FRAME_SIZE_SAMPLES>, | ||
| float, GSMFullRateCompressorInternal::GSM_SAMPLE_RATE>>; | ||
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| inline void init_gsm_full_rate_compressor(py::module &m) { | ||
| py::class_<GSMFullRateCompressor, Plugin>( | ||
| m, "GSMFullRateCompressor", | ||
| "Apply an GSM Full Rate compressor to emulate the sound of a GSM Full " | ||
| "Rate (\"2G\") cellular " | ||
| "phone connection. This plugin internally resamples the input audio to " | ||
| "8kHz.") | ||
| .def(py::init([](ResamplingQuality quality) { | ||
| auto plugin = std::make_unique<GSMFullRateCompressor>(); | ||
| plugin->getNestedPlugin().setQuality(quality); | ||
| return plugin; | ||
| }), | ||
| py::arg("quality") = ResamplingQuality::WindowedSinc) | ||
| .def("__repr__", | ||
| [](const GSMFullRateCompressor &plugin) { | ||
| std::ostringstream ss; | ||
| ss << "<pedalboard.GSMFullRateCompressor"; | ||
| ss << " at " << &plugin; | ||
| ss << ">"; | ||
| return ss.str(); | ||
| }) | ||
| .def_property( | ||
| "quality", | ||
| [](GSMFullRateCompressor &plugin) { | ||
| return plugin.getNestedPlugin().getQuality(); | ||
| }, | ||
| [](GSMFullRateCompressor &plugin, ResamplingQuality quality) { | ||
| return plugin.getNestedPlugin().setQuality(quality); | ||
| }); | ||
| } | ||
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| }; // namespace Pedalboard |
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -0,0 +1,86 @@ | ||
| #! /usr/bin/env python | ||
| # | ||
| # Copyright 2021 Spotify AB | ||
| # | ||
| # Licensed under the GNU Public License, Version 3.0 (the "License"); | ||
| # you may not use this file except in compliance with the License. | ||
| # You may obtain a copy of the License at | ||
| # | ||
| # https://www.gnu.org/licenses/gpl-3.0.html | ||
| # | ||
| # Unless required by applicable law or agreed to in writing, software | ||
| # distributed under the License is distributed on an "AS IS" BASIS, | ||
| # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| # See the License for the specific language governing permissions and | ||
| # limitations under the License. | ||
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| import pytest | ||
| import numpy as np | ||
| from pedalboard import GSMFullRateCompressor, Resample | ||
| from .utils import generate_sine_at | ||
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| # GSM is a _very_ lossy codec: | ||
| GSM_ABSOLUTE_TOLERANCE = 0.75 | ||
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| # Passing in a full-scale sine wave seems to often make GSM clip: | ||
| SINE_WAVE_VOLUME = 0.9 | ||
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| @pytest.mark.parametrize("fundamental_hz", [440.0]) | ||
| @pytest.mark.parametrize("sample_rate", [8000, 11025, 32001.2345, 44100, 48000]) | ||
| @pytest.mark.parametrize("buffer_size", [1, 32, 160, 1_000_000]) | ||
| @pytest.mark.parametrize("duration", [1.0]) | ||
| @pytest.mark.parametrize( | ||
| "quality", | ||
| [ | ||
| Resample.Quality.ZeroOrderHold, | ||
| Resample.Quality.Linear, | ||
| Resample.Quality.Lagrange, | ||
| Resample.Quality.CatmullRom, | ||
| Resample.Quality.WindowedSinc, | ||
| ], | ||
| ) | ||
| @pytest.mark.parametrize("num_channels", [1, 2]) | ||
| def test_gsm_compressor( | ||
| fundamental_hz: float, | ||
| sample_rate: float, | ||
| buffer_size: int, | ||
| duration: float, | ||
| quality: Resample.Quality, | ||
| num_channels: int, | ||
| ): | ||
| signal = ( | ||
| generate_sine_at(sample_rate, fundamental_hz, duration, num_channels) * SINE_WAVE_VOLUME | ||
| ) | ||
| output = GSMFullRateCompressor(quality=quality)(signal, sample_rate, buffer_size=buffer_size) | ||
| np.testing.assert_allclose(signal, output, atol=GSM_ABSOLUTE_TOLERANCE) | ||
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| @pytest.mark.parametrize("sample_rate", [8000, 44100]) | ||
| @pytest.mark.parametrize( | ||
| "quality", | ||
| [ | ||
| Resample.Quality.ZeroOrderHold, | ||
| Resample.Quality.Linear, | ||
| Resample.Quality.Lagrange, | ||
| Resample.Quality.CatmullRom, | ||
| Resample.Quality.WindowedSinc, | ||
| ], | ||
| ) | ||
| @pytest.mark.parametrize("num_channels", [1, 2]) | ||
| def test_gsm_compressor_invariant_to_buffer_size( | ||
| sample_rate: float, | ||
| quality: Resample.Quality, | ||
| num_channels: int, | ||
| ): | ||
| fundamental_hz = 400.0 | ||
| duration = 3.0 | ||
| signal = generate_sine_at(sample_rate, fundamental_hz, duration, num_channels) | ||
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| compressed = [ | ||
| GSMFullRateCompressor(quality=quality)(signal, sample_rate, buffer_size=buffer_size) | ||
| for buffer_size in (1, 32, 7000, 8192) | ||
| ] | ||
| for a, b in zip(compressed, compressed[1:]): | ||
| np.testing.assert_allclose(a, b) |
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