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https://github.com/LadybirdBrowser/ladybird.git
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Everywhere: rename 'Sample' type to 'Frame'
Because it's what it really is. A frame is composed of 1 or more samples, in the case of SerenityOS 2 (stereo). This will make it less confusing for future mantainability.
This commit is contained in:
parent
f4bd095aa3
commit
0d5e1e9df1
Notes:
sideshowbarker
2024-07-18 21:03:26 +09:00
Author: https://github.com/ngc6302h Commit: https://github.com/SerenityOS/serenity/commit/0d5e1e9df18 Pull-request: https://github.com/SerenityOS/serenity/pull/5878 Reviewed-by: https://github.com/Dexesttp Reviewed-by: https://github.com/RealKC Reviewed-by: https://github.com/alimpfard
@ -83,7 +83,7 @@ constexpr const char* wave_strings[] = {
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"Square",
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"Saw",
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"Noise",
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"Sample",
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"Frame",
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};
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constexpr int first_wave = Sine;
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@ -45,7 +45,7 @@ Track::~Track()
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void Track::fill_sample(Sample& sample)
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{
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Audio::Sample new_sample;
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Audio::Frame new_sample;
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for (size_t note = 0; note < note_count; ++note) {
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if (!m_roll_iters[note].is_end()) {
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@ -86,7 +86,7 @@ void Track::fill_sample(Sample& sample)
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VERIFY_NOT_REACHED();
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}
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Audio::Sample note_sample;
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Audio::Frame note_sample;
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switch (m_wave) {
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case Wave::Sine:
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note_sample = sine(note);
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@ -172,7 +172,7 @@ String Track::set_recorded_sample(const StringView& path)
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// All of the information for these waves is on Wikipedia.
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Audio::Sample Track::sine(size_t note)
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Audio::Frame Track::sine(size_t note)
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{
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double pos = note_frequencies[note] / sample_rate;
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double sin_step = pos * 2 * M_PI;
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@ -181,7 +181,7 @@ Audio::Sample Track::sine(size_t note)
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return w;
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}
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Audio::Sample Track::saw(size_t note)
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Audio::Frame Track::saw(size_t note)
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{
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double saw_step = note_frequencies[note] / sample_rate;
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double t = m_pos[note];
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@ -190,7 +190,7 @@ Audio::Sample Track::saw(size_t note)
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return w;
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}
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Audio::Sample Track::square(size_t note)
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Audio::Frame Track::square(size_t note)
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{
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double pos = note_frequencies[note] / sample_rate;
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double square_step = pos * 2 * M_PI;
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@ -199,7 +199,7 @@ Audio::Sample Track::square(size_t note)
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return w;
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}
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Audio::Sample Track::triangle(size_t note)
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Audio::Frame Track::triangle(size_t note)
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{
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double triangle_step = note_frequencies[note] / sample_rate;
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double t = m_pos[note];
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@ -208,14 +208,14 @@ Audio::Sample Track::triangle(size_t note)
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return w;
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}
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Audio::Sample Track::noise() const
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Audio::Frame Track::noise() const
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{
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double random_percentage = static_cast<double>(rand()) / RAND_MAX;
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double w = (random_percentage * 2) - 1;
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return w;
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}
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Audio::Sample Track::recorded_sample(size_t note)
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Audio::Frame Track::recorded_sample(size_t note)
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{
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int t = m_pos[note];
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if (t >= static_cast<int>(m_recorded_sample.size()))
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@ -42,7 +42,7 @@ public:
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explicit Track(const u32& time);
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~Track();
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const Vector<Audio::Sample>& recorded_sample() const { return m_recorded_sample; }
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const Vector<Audio::Frame>& recorded_sample() const { return m_recorded_sample; }
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const SinglyLinkedList<RollNote>& roll_notes(int note) const { return m_roll_notes[note]; }
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int wave() const { return m_wave; }
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int attack() const { return m_attack; }
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@ -65,19 +65,19 @@ public:
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void set_delay(int delay);
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private:
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Audio::Sample sine(size_t note);
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Audio::Sample saw(size_t note);
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Audio::Sample square(size_t note);
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Audio::Sample triangle(size_t note);
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Audio::Sample noise() const;
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Audio::Sample recorded_sample(size_t note);
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Audio::Frame sine(size_t note);
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Audio::Frame saw(size_t note);
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Audio::Frame square(size_t note);
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Audio::Frame triangle(size_t note);
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Audio::Frame noise() const;
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Audio::Frame recorded_sample(size_t note);
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void sync_roll(int note);
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void set_sustain_impl(int sustain);
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Vector<Sample> m_delay_buffer;
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Vector<Audio::Sample> m_recorded_sample;
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Vector<Audio::Frame> m_recorded_sample;
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u8 m_note_on[note_count] { 0 };
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double m_power[note_count] { 0 };
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@ -36,7 +36,7 @@ i32 Buffer::allocate_id()
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}
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template<typename SampleReader>
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static void read_samples_from_stream(InputMemoryStream& stream, SampleReader read_sample, Vector<Sample>& samples, ResampleHelper& resampler, int num_channels)
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static void read_samples_from_stream(InputMemoryStream& stream, SampleReader read_sample, Vector<Frame>& samples, ResampleHelper& resampler, int num_channels)
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{
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double norm_l = 0;
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double norm_r = 0;
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@ -45,7 +45,7 @@ static void read_samples_from_stream(InputMemoryStream& stream, SampleReader rea
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case 1:
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for (;;) {
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while (resampler.read_sample(norm_l, norm_r)) {
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samples.append(Sample(norm_l));
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samples.append(Frame(norm_l));
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}
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norm_l = read_sample(stream);
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@ -58,7 +58,7 @@ static void read_samples_from_stream(InputMemoryStream& stream, SampleReader rea
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case 2:
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for (;;) {
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while (resampler.read_sample(norm_l, norm_r)) {
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samples.append(Sample(norm_l, norm_r));
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samples.append(Frame(norm_l, norm_r));
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}
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norm_l = read_sample(stream);
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norm_r = read_sample(stream);
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@ -113,7 +113,7 @@ RefPtr<Buffer> Buffer::from_pcm_data(ReadonlyBytes data, ResampleHelper& resampl
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RefPtr<Buffer> Buffer::from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, int bits_per_sample, int num_samples)
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{
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Vector<Sample> fdata;
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Vector<Frame> fdata;
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fdata.ensure_capacity(num_samples);
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switch (bits_per_sample) {
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@ -37,22 +37,22 @@ 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 Sample {
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Sample()
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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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Sample(double left)
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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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Sample(double left, double right)
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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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@ -78,7 +78,7 @@ struct Sample {
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right *= pct;
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}
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Sample& operator+=(const Sample& other)
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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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@ -111,7 +111,7 @@ 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, int bits_per_sample);
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static RefPtr<Buffer> from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, int bits_per_sample, int num_samples);
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static NonnullRefPtr<Buffer> create_with_samples(Vector<Sample>&& samples)
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static NonnullRefPtr<Buffer> create_with_samples(Vector<Frame>&& samples)
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{
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return adopt(*new Buffer(move(samples)));
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}
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@ -120,20 +120,20 @@ public:
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return adopt(*new Buffer(move(buffer), buffer_id, sample_count));
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}
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const Sample* samples() const { return (const Sample*)data(); }
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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(Sample); }
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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<Sample> samples)
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: m_buffer(Core::AnonymousBuffer::create_with_size(samples.size() * sizeof(Sample)))
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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(Sample));
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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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active_mix_queues.remove_all_matching([&](auto& entry) { return !entry->client(); });
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Audio::Sample mixed_buffer[1024];
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auto mixed_buffer_length = (int)(sizeof(mixed_buffer) / sizeof(Audio::Sample));
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Audio::Frame mixed_buffer[1024];
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auto mixed_buffer_length = (int)(sizeof(mixed_buffer) / sizeof(Audio::Frame));
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// Mix the buffers together into the output
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for (auto& queue : active_mix_queues) {
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@ -98,7 +98,7 @@ void Mixer::mix()
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for (int i = 0; i < mixed_buffer_length; ++i) {
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auto& mixed_sample = mixed_buffer[i];
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Audio::Sample sample;
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Audio::Frame sample;
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if (!queue->get_next_sample(sample))
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break;
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mixed_sample += sample;
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@ -51,7 +51,7 @@ public:
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bool is_full() const { return m_queue.size() >= 3; }
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void enqueue(NonnullRefPtr<Audio::Buffer>&&);
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bool get_next_sample(Audio::Sample& sample)
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bool get_next_sample(Audio::Frame& sample)
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{
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if (m_paused)
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return false;
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