LibAudio: Rename Audio::Frame -> Audio::Sample

"Frame" is an MPEG term, which is not only unintuitive but also
overloaded with different meaning by other codecs (e.g. FLAC).
Therefore, use the standard term Sample for the central audio structure.

The class is also extracted to its own file, because it's becoming quite
large. Bundling these two changes means not distributing similar
modifications (changing names and paths) across commits.

Co-authored-by: kleines Filmröllchen <malu.bertsch@gmail.com>

11 files changed, 179 insertions, 165 deletions

diff --git a/Userland/Applications/Piano/AudioPlayerLoop.cpp b/Userland/Applications/Piano/AudioPlayerLoop.cpp
index 9a0861a0ed..dc19c69542 100644
--- a/Userland/Applications/Piano/AudioPlayerLoop.cpp
+++ b/Userland/Applications/Piano/AudioPlayerLoop.cpp
@@ -12,10 +12,10 @@
 // Converts Piano-internal data to an Audio::Buffer that AudioServer receives
 static NonnullRefPtr<Audio::Buffer> music_samples_to_buffer(Array<Sample, sample_count> samples)
 {
-    Vector<Audio::Frame, sample_count> frames;
+    Vector<Audio::Sample, sample_count> frames;
     frames.ensure_capacity(sample_count);
     for (auto sample : samples) {
-        Audio::Frame frame = { sample.left / (double)NumericLimits<i16>::max(), sample.right / (double)NumericLimits<i16>::max() };
+        Audio::Sample frame = { sample.left / (double)NumericLimits<i16>::max(), sample.right / (double)NumericLimits<i16>::max() };
         frames.unchecked_append(frame);
     }
     return Audio::Buffer::create_with_samples(frames);
diff --git a/Userland/Applications/Piano/Track.cpp b/Userland/Applications/Piano/Track.cpp
index 1cd6b7be71..4fcba80f5a 100644
--- a/Userland/Applications/Piano/Track.cpp
+++ b/Userland/Applications/Piano/Track.cpp
@@ -31,7 +31,7 @@ Track::~Track()
 
 void Track::fill_sample(Sample& sample)
 {
-    Audio::Frame new_sample;
+    Audio::Sample new_sample;
 
     for (size_t note = 0; note < note_count; ++note) {
         if (!m_roll_iterators[note].is_end()) {
@@ -72,7 +72,7 @@ void Track::fill_sample(Sample& sample)
             VERIFY_NOT_REACHED();
         }
 
-        Audio::Frame note_sample;
+        Audio::Sample note_sample;
         switch (m_wave) {
         case Wave::Sine:
             note_sample = sine(note);
@@ -161,7 +161,7 @@ String Track::set_recorded_sample(const StringView& path)
 
 // All of the information for these waves is on Wikipedia.
 
-Audio::Frame Track::sine(size_t note)
+Audio::Sample Track::sine(size_t note)
 {
     double pos = note_frequencies[note] / sample_rate;
     double sin_step = pos * 2 * M_PI;
@@ -170,7 +170,7 @@ Audio::Frame Track::sine(size_t note)
     return w;
 }
 
-Audio::Frame Track::saw(size_t note)
+Audio::Sample Track::saw(size_t note)
 {
     double saw_step = note_frequencies[note] / sample_rate;
     double t = m_pos[note];
@@ -179,7 +179,7 @@ Audio::Frame Track::saw(size_t note)
     return w;
 }
 
-Audio::Frame Track::square(size_t note)
+Audio::Sample Track::square(size_t note)
 {
     double pos = note_frequencies[note] / sample_rate;
     double square_step = pos * 2 * M_PI;
@@ -188,7 +188,7 @@ Audio::Frame Track::square(size_t note)
     return w;
 }
 
-Audio::Frame Track::triangle(size_t note)
+Audio::Sample Track::triangle(size_t note)
 {
     double triangle_step = note_frequencies[note] / sample_rate;
     double t = m_pos[note];
@@ -197,7 +197,7 @@ Audio::Frame Track::triangle(size_t note)
     return w;
 }
 
-Audio::Frame Track::noise(size_t note)
+Audio::Sample Track::noise(size_t note)
 {
     double step = note_frequencies[note] / sample_rate;
     // m_pos keeps track of the time since the last random sample
@@ -210,7 +210,7 @@ Audio::Frame Track::noise(size_t note)
     return m_last_w[note];
 }
 
-Audio::Frame Track::recorded_sample(size_t note)
+Audio::Sample Track::recorded_sample(size_t note)
 {
     int t = m_pos[note];
     if (t >= static_cast<int>(m_recorded_sample.size()))
diff --git a/Userland/Applications/Piano/Track.h b/Userland/Applications/Piano/Track.h
index 42cf32114d..2397cd776f 100644
--- a/Userland/Applications/Piano/Track.h
+++ b/Userland/Applications/Piano/Track.h
@@ -24,7 +24,7 @@ public:
     explicit Track(const u32& time);
     ~Track();
 
-    const Vector<Audio::Frame>& recorded_sample() const { return m_recorded_sample; }
+    const Vector<Audio::Sample>& recorded_sample() const { return m_recorded_sample; }
     const SinglyLinkedList<RollNote>& roll_notes(int note) const { return m_roll_notes[note]; }
     int wave() const { return m_wave; }
     int volume() const { return m_volume; }
@@ -48,17 +48,17 @@ public:
     void set_release(int release);
 
 private:
-    Audio::Frame sine(size_t note);
-    Audio::Frame saw(size_t note);
-    Audio::Frame square(size_t note);
-    Audio::Frame triangle(size_t note);
-    Audio::Frame noise(size_t note);
-    Audio::Frame recorded_sample(size_t note);
+    Audio::Sample sine(size_t note);
+    Audio::Sample saw(size_t note);
+    Audio::Sample square(size_t note);
+    Audio::Sample triangle(size_t note);
+    Audio::Sample noise(size_t note);
+    Audio::Sample recorded_sample(size_t note);
 
     void sync_roll(int note);
     void set_sustain_impl(int sustain);
 
-    Vector<Audio::Frame> m_recorded_sample;
+    Vector<Audio::Sample> m_recorded_sample;
 
     u8 m_note_on[note_count] { 0 };
     double m_power[note_count] { 0 };
diff --git a/Userland/Libraries/LibAudio/Buffer.cpp b/Userland/Libraries/LibAudio/Buffer.cpp
index 4f401b4f28..b53fc9d5bf 100644
--- a/Userland/Libraries/LibAudio/Buffer.cpp
+++ b/Userland/Libraries/LibAudio/Buffer.cpp
@@ -45,7 +45,7 @@ i32 Buffer::allocate_id()
 }
 
 template<typename SampleReader>
-static void read_samples_from_stream(InputMemoryStream& stream, SampleReader read_sample, Vector<Frame>& samples, int num_channels)
+static void read_samples_from_stream(InputMemoryStream& stream, SampleReader read_sample, Vector<Sample>& samples, int num_channels)
 {
     double norm_l = 0;
     double norm_r = 0;
@@ -54,7 +54,7 @@ static void read_samples_from_stream(InputMemoryStream& stream, SampleReader rea
     case 1:
         for (;;) {
             norm_l = read_sample(stream);
-            samples.append(Frame(norm_l));
+            samples.append(Sample(norm_l));
 
             if (stream.handle_any_error()) {
                 break;
@@ -65,7 +65,7 @@ static void read_samples_from_stream(InputMemoryStream& stream, SampleReader rea
         for (;;) {
             norm_l = read_sample(stream);
             norm_r = read_sample(stream);
-            samples.append(Frame(norm_l, norm_r));
+            samples.append(Sample(norm_l, norm_r));
 
             if (stream.handle_any_error()) {
                 break;
@@ -130,7 +130,7 @@ RefPtr<Buffer> Buffer::from_pcm_data(ReadonlyBytes data, int num_channels, PcmSa
 
 RefPtr<Buffer> Buffer::from_pcm_stream(InputMemoryStream& stream, int num_channels, PcmSampleFormat sample_format, int num_samples)
 {
-    Vector<Frame> fdata;
+    Vector<Sample> fdata;
     fdata.ensure_capacity(num_samples);
 
     switch (sample_format) {
@@ -189,7 +189,7 @@ template Vector<double> ResampleHelper<double>::resample(Vector<double>);
 
 NonnullRefPtr<Buffer> resample_buffer(ResampleHelper<double>& resampler, Buffer const& to_resample)
 {
-    Vector<Frame> resampled;
+    Vector<Sample> resampled;
     resampled.ensure_capacity(to_resample.sample_count() * ceil_div(resampler.source(), resampler.target()));
     for (size_t i = 0; i < static_cast<size_t>(to_resample.sample_count()); ++i) {
         auto sample = to_resample.samples()[i];
diff --git a/Userland/Libraries/LibAudio/Buffer.h b/Userland/Libraries/LibAudio/Buffer.h
index 6020a0f6a5..dbc08814e6 100644
--- a/Userland/Libraries/LibAudio/Buffer.h
+++ b/Userland/Libraries/LibAudio/Buffer.h
@@ -8,144 +8,17 @@
 #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 <LibAudio/Sample.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() = default;
-
-    // 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).
-    //
-    // Format ranges:
-    // - Linear:        0.0 to 1.0
-    // - Logarithmic:   0.0 to 1.0
-
-    ALWAYS_INLINE double linear_to_log(double const change)
-    {
-        // TODO: Add linear slope around 0
-        return VOLUME_A * exp(VOLUME_B * change);
-    }
-
-    ALWAYS_INLINE double log_to_linear(double const val)
-    {
-        // TODO: Add linear slope around 0
-        return log(val / VOLUME_A) / VOLUME_B;
-    }
-
-    ALWAYS_INLINE Frame& log_multiply(double const change)
-    {
-        double factor = linear_to_log(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;
-    }
-
-    ALWAYS_INLINE Frame& log_pan(double const pan)
-    {
-        left *= linear_to_log(min(pan * -1 + 1.0, 1.0));
-        right *= linear_to_log(min(pan + 1.0, 1.0));
-        return *this;
-    }
-
-    ALWAYS_INLINE Frame log_pan(double const pan) const
-    {
-        Frame new_frame { left, right };
-        new_frame.log_pan(pan);
-        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+=(double other)
-    {
-        left += other;
-        right += other;
-        return *this;
-    }
-
-    constexpr Frame operator+(Frame const& other)
-    {
-        return { left + other.left, right + other.right };
-    }
-
-    double left { 0 };
-    double right { 0 };
-};
-
 // Supported PCM sample formats.
 enum PcmSampleFormat : u8 {
     Uint8,
@@ -196,7 +69,7 @@ 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)
+    static NonnullRefPtr<Buffer> create_with_samples(Vector<Sample>&& samples)
     {
         return adopt_ref(*new Buffer(move(samples)));
     }
@@ -205,20 +78,20 @@ public:
         return adopt_ref(*new Buffer(move(buffer), buffer_id, sample_count));
     }
 
-    const Frame* samples() const { return (const Frame*)data(); }
+    const Sample* samples() const { return (const Sample*)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 size_in_bytes() const { return m_sample_count * (int)sizeof(Sample); }
     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)).release_value())
+    explicit Buffer(const Vector<Sample> samples)
+        : m_buffer(Core::AnonymousBuffer::create_with_size(samples.size() * sizeof(Sample)).release_value())
         , m_id(allocate_id())
         , m_sample_count(samples.size())
     {
-        memcpy(m_buffer.data<void>(), samples.data(), samples.size() * sizeof(Frame));
+        memcpy(m_buffer.data<void>(), samples.data(), samples.size() * sizeof(Sample));
     }
 
     explicit Buffer(Core::AnonymousBuffer buffer, i32 buffer_id, int sample_count)
diff --git a/Userland/Libraries/LibAudio/FlacLoader.cpp b/Userland/Libraries/LibAudio/FlacLoader.cpp
index f3c061ec94..02aa809b24 100644
--- a/Userland/Libraries/LibAudio/FlacLoader.cpp
+++ b/Userland/Libraries/LibAudio/FlacLoader.cpp
@@ -231,7 +231,7 @@ void FlacLoaderPlugin::seek(const int position)
 
 RefPtr<Buffer> FlacLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_input)
 {
-    Vector<Frame> samples;
+    Vector<Sample> samples;
     ssize_t remaining_samples = m_total_samples - m_loaded_samples;
     if (remaining_samples <= 0) {
         return nullptr;
@@ -417,7 +417,7 @@ void FlacLoaderPlugin::next_frame()
     m_current_frame_data.ensure_capacity(left.size());
     // zip together channels
     for (size_t i = 0; i < left.size(); ++i) {
-        Frame frame = { left[i] / sample_rescale, right[i] / sample_rescale };
+        Sample frame = { left[i] / sample_rescale, right[i] / sample_rescale };
         m_current_frame_data.unchecked_append(frame);
     }
 
diff --git a/Userland/Libraries/LibAudio/FlacLoader.h b/Userland/Libraries/LibAudio/FlacLoader.h
index 915bd44b5c..f80fc57f65 100644
--- a/Userland/Libraries/LibAudio/FlacLoader.h
+++ b/Userland/Libraries/LibAudio/FlacLoader.h
@@ -143,7 +143,7 @@ private:
     u64 m_data_start_location { 0 };
     OwnPtr<FlacInputStream> m_stream;
     Optional<FlacFrameHeader> m_current_frame;
-    Vector<Frame> m_current_frame_data;
+    Vector<Sample> m_current_frame_data;
     u64 m_current_sample_or_frame { 0 };
 };
 
diff --git a/Userland/Libraries/LibAudio/Sample.h b/Userland/Libraries/LibAudio/Sample.h
new file mode 100644
index 0000000000..23faf6244a
--- /dev/null
+++ b/Userland/Libraries/LibAudio/Sample.h
@@ -0,0 +1,141 @@
+/*
+ * 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/Math.h>
+
+namespace Audio {
+using namespace AK::Exponentials;
+// Constants for logarithmic volume. See Sample::linear_to_log
+// 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 Sample {
+    constexpr Sample() = default;
+
+    // For mono
+    constexpr Sample(double left)
+        : left(left)
+        , right(left)
+    {
+    }
+
+    // For stereo
+    constexpr Sample(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).
+    //
+    // Format ranges:
+    // - Linear:        0.0 to 1.0
+    // - Logarithmic:   0.0 to 1.0
+
+    ALWAYS_INLINE double linear_to_log(double const change)
+    {
+        // TODO: Add linear slope around 0
+        return VOLUME_A * exp(VOLUME_B * change);
+    }
+
+    ALWAYS_INLINE double log_to_linear(double const val)
+    {
+        // TODO: Add linear slope around 0
+        return log(val / VOLUME_A) / VOLUME_B;
+    }
+
+    ALWAYS_INLINE Sample& log_multiply(double const change)
+    {
+        double factor = linear_to_log(change);
+        left *= factor;
+        right *= factor;
+        return *this;
+    }
+
+    ALWAYS_INLINE Sample log_multiplied(double const volume_change) const
+    {
+        Sample new_frame { left, right };
+        new_frame.log_multiply(volume_change);
+        return new_frame;
+    }
+
+    ALWAYS_INLINE Sample& log_pan(double const pan)
+    {
+        left *= linear_to_log(min(pan * -1 + 1.0, 1.0));
+        right *= linear_to_log(min(pan + 1.0, 1.0));
+        return *this;
+    }
+
+    ALWAYS_INLINE Sample log_pan(double const pan) const
+    {
+        Sample new_frame { left, right };
+        new_frame.log_pan(pan);
+        return new_frame;
+    }
+
+    constexpr Sample& operator*=(double const mult)
+    {
+        left *= mult;
+        right *= mult;
+        return *this;
+    }
+
+    constexpr Sample operator*(double const mult)
+    {
+        return { left * mult, right * mult };
+    }
+
+    constexpr Sample& operator+=(Sample const& other)
+    {
+        left += other.left;
+        right += other.right;
+        return *this;
+    }
+    constexpr Sample& operator+=(double other)
+    {
+        left += other;
+        right += other;
+        return *this;
+    }
+
+    constexpr Sample operator+(Sample const& other)
+    {
+        return { left + other.left, right + other.right };
+    }
+
+    double left { 0 };
+    double right { 0 };
+};
+
+}
diff --git a/Userland/Libraries/LibDSP/Music.h b/Userland/Libraries/LibDSP/Music.h
index 5a9c434a6b..306b3c6193 100644
--- a/Userland/Libraries/LibDSP/Music.h
+++ b/Userland/Libraries/LibDSP/Music.h
@@ -14,7 +14,7 @@
 namespace LibDSP {
 
 // FIXME: Audio::Frame is 64-bit float, which is quite large for long clips.
-using Sample = Audio::Frame;
+using Sample = Audio::Sample;
 
 Sample const SAMPLE_OFF = { 0.0, 0.0 };
 
diff --git a/Userland/Services/AudioServer/Mixer.cpp b/Userland/Services/AudioServer/Mixer.cpp
index bc57b25eda..99d6038930 100644
--- a/Userland/Services/AudioServer/Mixer.cpp
+++ b/Userland/Services/AudioServer/Mixer.cpp
@@ -77,8 +77,8 @@ void Mixer::mix()
 
         active_mix_queues.remove_all_matching([&](auto& entry) { return !entry->client(); });
 
-        Audio::Frame mixed_buffer[1024];
-        auto mixed_buffer_length = (int)(sizeof(mixed_buffer) / sizeof(Audio::Frame));
+        Audio::Sample mixed_buffer[1024];
+        auto mixed_buffer_length = (int)(sizeof(mixed_buffer) / sizeof(Audio::Sample));
 
         m_main_volume.advance_time();
 
@@ -94,7 +94,7 @@ void Mixer::mix()
 
             for (int i = 0; i < mixed_buffer_length; ++i) {
                 auto& mixed_sample = mixed_buffer[i];
-                Audio::Frame sample;
+                Audio::Sample sample;
                 if (!queue->get_next_sample(sample))
                     break;
                 sample.log_multiply(SAMPLE_HEADROOM);
diff --git a/Userland/Services/AudioServer/Mixer.h b/Userland/Services/AudioServer/Mixer.h
index a9dbde4cc8..0cd48c8500 100644
--- a/Userland/Services/AudioServer/Mixer.h
+++ b/Userland/Services/AudioServer/Mixer.h
@@ -40,7 +40,7 @@ public:
     bool is_full() const { return m_queue.size() >= 3; }
     void enqueue(NonnullRefPtr<Audio::Buffer>&&);
 
-    bool get_next_sample(Audio::Frame& sample)
+    bool get_next_sample(Audio::Sample& sample)
     {
         if (m_paused)
             return false;