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LibAudio's WavLoader plugin for loading WAV files now supports loading audio files with 32-bit float or 64-bit float samples. By supporting these new non-int sample formats, Audio::Buffer now stores the sample format (out of a list of supported formats) instead of the raw bit depth. (The bit depth is easily calculated with pcm_bits_per_sample)
148 lines
3.8 KiB
C++
148 lines
3.8 KiB
C++
/*
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* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
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* Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#pragma once
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#include <AK/ByteBuffer.h>
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#include <AK/MemoryStream.h>
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#include <AK/String.h>
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#include <AK/Types.h>
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#include <AK/Vector.h>
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#include <LibCore/AnonymousBuffer.h>
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#include <string.h>
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namespace Audio {
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// A single sample in an audio buffer.
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// Values are floating point, and should range from -1.0 to +1.0
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struct Frame {
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Frame()
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: left(0)
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, right(0)
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{
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}
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// For mono
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Frame(double left)
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: left(left)
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, right(left)
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{
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}
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// For stereo
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Frame(double left, double right)
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: left(left)
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, right(right)
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{
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}
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void clip()
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{
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if (left > 1)
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left = 1;
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else if (left < -1)
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left = -1;
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if (right > 1)
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right = 1;
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else if (right < -1)
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right = -1;
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}
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void scale(int percent)
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{
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double pct = (double)percent / 100.0;
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left *= pct;
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right *= pct;
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}
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Frame& operator+=(const Frame& other)
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{
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left += other.left;
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right += other.right;
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return *this;
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}
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double left;
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double right;
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};
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// Supported PCM sample formats.
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enum PcmSampleFormat : u8 {
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Uint8,
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Int16,
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Int24,
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Float32,
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Float64,
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};
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// Most of the read code only cares about how many bits to read or write
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u16 pcm_bits_per_sample(PcmSampleFormat format);
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String sample_format_name(PcmSampleFormat format);
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// Small helper to resample from one playback rate to another
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// This isn't really "smart", in that we just insert (or drop) samples.
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// Should do better...
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class ResampleHelper {
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public:
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ResampleHelper(double source, double target);
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void process_sample(double sample_l, double sample_r);
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bool read_sample(double& next_l, double& next_r);
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private:
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const double m_ratio;
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double m_current_ratio { 0 };
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double m_last_sample_l { 0 };
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double m_last_sample_r { 0 };
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};
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// A buffer of audio samples, normalized to 44100hz.
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class Buffer : public RefCounted<Buffer> {
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public:
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static RefPtr<Buffer> from_pcm_data(ReadonlyBytes data, ResampleHelper& resampler, int num_channels, PcmSampleFormat sample_format);
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static RefPtr<Buffer> from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, PcmSampleFormat sample_format, int num_samples);
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static NonnullRefPtr<Buffer> create_with_samples(Vector<Frame>&& samples)
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{
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return adopt_ref(*new Buffer(move(samples)));
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}
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static NonnullRefPtr<Buffer> create_with_anonymous_buffer(Core::AnonymousBuffer buffer, i32 buffer_id, int sample_count)
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{
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return adopt_ref(*new Buffer(move(buffer), buffer_id, sample_count));
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}
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const Frame* samples() const { return (const Frame*)data(); }
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int sample_count() const { return m_sample_count; }
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const void* data() const { return m_buffer.data<void>(); }
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int size_in_bytes() const { return m_sample_count * (int)sizeof(Frame); }
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int id() const { return m_id; }
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const Core::AnonymousBuffer& anonymous_buffer() const { return m_buffer; }
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private:
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explicit Buffer(const Vector<Frame> samples)
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: m_buffer(Core::AnonymousBuffer::create_with_size(samples.size() * sizeof(Frame)))
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, m_id(allocate_id())
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, m_sample_count(samples.size())
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{
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memcpy(m_buffer.data<void>(), samples.data(), samples.size() * sizeof(Frame));
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}
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explicit Buffer(Core::AnonymousBuffer buffer, i32 buffer_id, int sample_count)
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: m_buffer(move(buffer))
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, m_id(buffer_id)
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, m_sample_count(sample_count)
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{
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}
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static i32 allocate_id();
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Core::AnonymousBuffer m_buffer;
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const i32 m_id;
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const int m_sample_count;
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};
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}
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