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serenity/Userland/Libraries/LibAudio/Buffer.h
kleines Filmröllchen ab4a2b8b41 LibDSP+LibAudio: Use logarithmic scaling in delay effect
With logarithmic volume scaling, the delay effect can sound more
natural.
2021-09-12 23:38:57 +02:00

208 lines
6 KiB
C++

/*
* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
* Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/ByteBuffer.h>
#include <AK/Math.h>
#include <AK/MemoryStream.h>
#include <AK/String.h>
#include <AK/Types.h>
#include <AK/Vector.h>
#include <LibCore/AnonymousBuffer.h>
#include <string.h>
namespace Audio {
using namespace AK::Exponentials;
// Constants for logarithmic volume. See Frame::operator*
// Corresponds to 60dB
constexpr double DYNAMIC_RANGE = 1000;
constexpr double VOLUME_A = 1 / DYNAMIC_RANGE;
double const VOLUME_B = log(DYNAMIC_RANGE);
// A single sample in an audio buffer.
// Values are floating point, and should range from -1.0 to +1.0
struct Frame {
constexpr Frame()
: left(0)
, right(0)
{
}
// For mono
constexpr Frame(double left)
: left(left)
, right(left)
{
}
// For stereo
constexpr Frame(double left, double right)
: left(left)
, right(right)
{
}
void clip()
{
if (left > 1)
left = 1;
else if (left < -1)
left = -1;
if (right > 1)
right = 1;
else if (right < -1)
right = -1;
}
// Logarithmic scaling, as audio should ALWAYS do.
// Reference: https://www.dr-lex.be/info-stuff/volumecontrols.html
// We use the curve `factor = a * exp(b * change)`,
// where change is the input fraction we want to change by,
// a = 1/1000, b = ln(1000) = 6.908 and factor is the multiplier used.
// The value 1000 represents the dynamic range in sound pressure, which corresponds to 60 dB(A).
// This is a good dynamic range because it can represent all loudness values from
// 30 dB(A) (barely hearable with background noise)
// to 90 dB(A) (almost too loud to hear and about the reasonable limit of actual sound equipment).
ALWAYS_INLINE Frame& log_multiply(double const change)
{
double factor = VOLUME_A * exp(VOLUME_B * change);
left *= factor;
right *= factor;
return *this;
}
ALWAYS_INLINE Frame log_multiplied(double const volume_change) const
{
Frame new_frame { left, right };
new_frame.log_multiply(volume_change);
return new_frame;
}
constexpr Frame& operator*=(double const mult)
{
left *= mult;
right *= mult;
return *this;
}
constexpr Frame operator*(double const mult)
{
return { left * mult, right * mult };
}
constexpr Frame& operator+=(Frame const& other)
{
left += other.left;
right += other.right;
return *this;
}
constexpr Frame operator+(Frame const& other)
{
return { left + other.left, right + other.right };
}
double left;
double right;
};
// Supported PCM sample formats.
enum PcmSampleFormat : u8 {
Uint8,
Int16,
Int24,
Int32,
Float32,
Float64,
};
// Most of the read code only cares about how many bits to read or write
u16 pcm_bits_per_sample(PcmSampleFormat format);
String sample_format_name(PcmSampleFormat format);
// Small helper to resample from one playback rate to another
// This isn't really "smart", in that we just insert (or drop) samples.
// Should do better...
template<typename SampleType>
class ResampleHelper {
public:
ResampleHelper(u32 source, u32 target);
// To be used as follows:
// while the resampler doesn't need a new sample, read_sample(current) and store the resulting samples.
// as long as the resampler needs a new sample, process_sample(current)
// Stores a new sample
void process_sample(SampleType sample_l, SampleType sample_r);
// Assigns the given sample to its correct value and returns false if there is a new sample required
bool read_sample(SampleType& next_l, SampleType& next_r);
Vector<SampleType> resample(Vector<SampleType> to_resample);
void reset();
u32 source() const { return m_source; }
u32 target() const { return m_target; }
private:
const u32 m_source;
const u32 m_target;
u32 m_current_ratio { 0 };
SampleType m_last_sample_l;
SampleType m_last_sample_r;
};
// A buffer of audio samples.
class Buffer : public RefCounted<Buffer> {
public:
static RefPtr<Buffer> from_pcm_data(ReadonlyBytes data, int num_channels, PcmSampleFormat sample_format);
static RefPtr<Buffer> from_pcm_stream(InputMemoryStream& stream, int num_channels, PcmSampleFormat sample_format, int num_samples);
static NonnullRefPtr<Buffer> create_with_samples(Vector<Frame>&& samples)
{
return adopt_ref(*new Buffer(move(samples)));
}
static NonnullRefPtr<Buffer> create_with_anonymous_buffer(Core::AnonymousBuffer buffer, i32 buffer_id, int sample_count)
{
return adopt_ref(*new Buffer(move(buffer), buffer_id, sample_count));
}
const Frame* samples() const { return (const Frame*)data(); }
int sample_count() const { return m_sample_count; }
const void* data() const { return m_buffer.data<void>(); }
int size_in_bytes() const { return m_sample_count * (int)sizeof(Frame); }
int id() const { return m_id; }
const Core::AnonymousBuffer& anonymous_buffer() const { return m_buffer; }
private:
explicit Buffer(const Vector<Frame> samples)
: m_buffer(Core::AnonymousBuffer::create_with_size(samples.size() * sizeof(Frame)))
, m_id(allocate_id())
, m_sample_count(samples.size())
{
memcpy(m_buffer.data<void>(), samples.data(), samples.size() * sizeof(Frame));
}
explicit Buffer(Core::AnonymousBuffer buffer, i32 buffer_id, int sample_count)
: m_buffer(move(buffer))
, m_id(buffer_id)
, m_sample_count(sample_count)
{
}
static i32 allocate_id();
Core::AnonymousBuffer m_buffer;
const i32 m_id;
const int m_sample_count;
};
// This only works for double resamplers, and therefore cannot be part of the class
NonnullRefPtr<Buffer> resample_buffer(ResampleHelper<double>& resampler, Buffer const& to_resample);
}