Audio: Fix code smells and issues found by static analysis

This fixes all current code smells, bugs and issues reported by
SonarCloud static analysis. Other issues are almost exclusively false
positives. This makes much code clearer, and some minor benefits in
performance or bug evasion may be gained.

12 files changed, 99 insertions, 77 deletions

diff --git a/Userland/Applications/Piano/Track.cpp b/Userland/Applications/Piano/Track.cpp
index 0743db4e6e..40899e4568 100644
--- a/Userland/Applications/Piano/Track.cpp
+++ b/Userland/Applications/Piano/Track.cpp
@@ -167,7 +167,7 @@ Audio::Sample Track::sine(size_t note)
     double sin_step = pos * 2 * M_PI;
     double w = sin(m_pos[note]);
     m_pos[note] += sin_step;
-    return w;
+    return Audio::Sample { w };
 }
 
 Audio::Sample Track::saw(size_t note)
@@ -176,7 +176,7 @@ Audio::Sample Track::saw(size_t note)
     double t = m_pos[note];
     double w = (0.5 - (t - floor(t))) * 2;
     m_pos[note] += saw_step;
-    return w;
+    return Audio::Sample { w };
 }
 
 Audio::Sample Track::square(size_t note)
@@ -185,7 +185,7 @@ Audio::Sample Track::square(size_t note)
     double square_step = pos * 2 * M_PI;
     double w = AK::sin(m_pos[note]) >= 0 ? 1 : -1;
     m_pos[note] += square_step;
-    return w;
+    return Audio::Sample { w };
 }
 
 Audio::Sample Track::triangle(size_t note)
@@ -194,7 +194,7 @@ Audio::Sample Track::triangle(size_t note)
     double t = m_pos[note];
     double w = AK::fabs(AK::fmod((4 * t) + 1, 4.) - 2) - 1.;
     m_pos[note] += triangle_step;
-    return w;
+    return Audio::Sample { w };
 }
 
 Audio::Sample Track::noise(size_t note)
@@ -207,14 +207,14 @@ Audio::Sample Track::noise(size_t note)
         m_last_w[note] = (random_percentage * 2) - 1;
         m_pos[note] = 0;
     }
-    return m_last_w[note];
+    return Audio::Sample { m_last_w[note] };
 }
 
 Audio::Sample Track::recorded_sample(size_t note)
 {
     int t = m_pos[note];
     if (t >= static_cast<int>(m_recorded_sample.size()))
-        return 0;
+        return {};
     double w_left = m_recorded_sample[t].left;
     double w_right = m_recorded_sample[t].right;
     if (t + 1 < static_cast<int>(m_recorded_sample.size())) {
diff --git a/Userland/Libraries/LibAudio/ClientConnection.cpp b/Userland/Libraries/LibAudio/ClientConnection.cpp
index 07d4312d5c..22581b1511 100644
--- a/Userland/Libraries/LibAudio/ClientConnection.cpp
+++ b/Userland/Libraries/LibAudio/ClientConnection.cpp
@@ -6,9 +6,14 @@
 
 #include <LibAudio/Buffer.h>
 #include <LibAudio/ClientConnection.h>
+#include <time.h>
 
 namespace Audio {
 
+// FIXME: We don't know what is a good value for this.
+// Real-time audio may be improved with a lower value.
+static timespec g_enqueue_wait_time { 0, 10'000'000 };
+
 ClientConnection::ClientConnection()
     : IPC::ServerConnection<AudioClientEndpoint, AudioServerEndpoint>(*this, "/tmp/portal/audio")
 {
@@ -20,9 +25,7 @@ void ClientConnection::enqueue(Buffer const& buffer)
         auto success = enqueue_buffer(buffer.anonymous_buffer(), buffer.id(), buffer.sample_count());
         if (success)
             break;
-        // FIXME: We don't know what is a good value for this.
-        // For now, decrease it to enable better real-time audio.
-        usleep(10000);
+        nanosleep(&g_enqueue_wait_time, nullptr);
     }
 }
 
diff --git a/Userland/Libraries/LibAudio/FlacLoader.cpp b/Userland/Libraries/LibAudio/FlacLoader.cpp
index e64c1fd028..af463fbebc 100644
--- a/Userland/Libraries/LibAudio/FlacLoader.cpp
+++ b/Userland/Libraries/LibAudio/FlacLoader.cpp
@@ -85,7 +85,7 @@ bool FlacLoaderPlugin::parse_header()
     } while (0)
 
     // Magic number
-    u32 flac = bit_input.read_bits_big_endian(32);
+    u32 flac = static_cast<u32>(bit_input.read_bits_big_endian(32));
     m_data_start_location += 4;
     ok = ok && flac == 0x664C6143; // "flaC"
     CHECK_OK("FLAC magic number");
@@ -102,20 +102,20 @@ bool FlacLoaderPlugin::parse_header()
     ScopeGuard clear_streaminfo_errors([&streaminfo_data] { streaminfo_data.handle_any_error(); });
 
     // STREAMINFO block
-    m_min_block_size = streaminfo_data.read_bits_big_endian(16);
+    m_min_block_size = static_cast<u16>(streaminfo_data.read_bits_big_endian(16));
     ok = ok && (m_min_block_size >= 16);
     CHECK_OK("Minimum block size");
-    m_max_block_size = streaminfo_data.read_bits_big_endian(16);
+    m_max_block_size = static_cast<u16>(streaminfo_data.read_bits_big_endian(16));
     ok = ok && (m_max_block_size >= 16);
     CHECK_OK("Maximum block size");
-    m_min_frame_size = streaminfo_data.read_bits_big_endian(24);
-    m_max_frame_size = streaminfo_data.read_bits_big_endian(24);
-    m_sample_rate = streaminfo_data.read_bits_big_endian(20);
+    m_min_frame_size = static_cast<u32>(streaminfo_data.read_bits_big_endian(24));
+    m_max_frame_size = static_cast<u32>(streaminfo_data.read_bits_big_endian(24));
+    m_sample_rate = static_cast<u32>(streaminfo_data.read_bits_big_endian(20));
     ok = ok && (m_sample_rate <= 655350);
     CHECK_OK("Sample rate");
-    m_num_channels = streaminfo_data.read_bits_big_endian(3) + 1; // 0 ^= one channel
+    m_num_channels = static_cast<u8>(streaminfo_data.read_bits_big_endian(3)) + 1; // 0 ^= one channel
 
-    u8 bits_per_sample = streaminfo_data.read_bits_big_endian(5) + 1;
+    u8 bits_per_sample = static_cast<u8>(streaminfo_data.read_bits_big_endian(5)) + 1;
     if (bits_per_sample == 8) {
         // FIXME: Signed/Unsigned issues?
         m_sample_format = PcmSampleFormat::Uint8;
@@ -130,7 +130,7 @@ bool FlacLoaderPlugin::parse_header()
         CHECK_OK("Sample bit depth");
     }
 
-    m_total_samples = streaminfo_data.read_bits_big_endian(36);
+    m_total_samples = static_cast<u64>(streaminfo_data.read_bits_big_endian(36));
     ok = ok && (m_total_samples > 0);
     CHECK_OK("Number of samples");
     // Parse checksum into a buffer first
@@ -164,7 +164,7 @@ bool FlacLoaderPlugin::parse_header()
         for (unsigned int i = 0; i < md5_checksum.size(); ++i) {
             checksum_string.appendff("{:0X}", md5_checksum[i]);
         }
-        dbgln("Parsed FLAC header: blocksize {}-{}{}, framesize {}-{}, {}Hz, {}bit, {} channels, {} samples total ({:.2f}s), MD5 {}, data start at {:x} bytes, {} headers total (skipped {})", m_min_block_size, m_max_block_size, is_fixed_blocksize_stream() ? " (constant)" : "", m_min_frame_size, m_max_frame_size, m_sample_rate, pcm_bits_per_sample(m_sample_format), m_num_channels, m_total_samples, m_total_samples / static_cast<double>(m_sample_rate), checksum_string.to_string(), m_data_start_location, total_meta_blocks, total_meta_blocks - meta_blocks_parsed);
+        dbgln("Parsed FLAC header: blocksize {}-{}{}, framesize {}-{}, {}Hz, {}bit, {} channels, {} samples total ({:.2f}s), MD5 {}, data start at {:x} bytes, {} headers total (skipped {})", m_min_block_size, m_max_block_size, is_fixed_blocksize_stream() ? " (constant)" : "", m_min_frame_size, m_max_frame_size, m_sample_rate, pcm_bits_per_sample(m_sample_format), m_num_channels, m_total_samples, static_cast<double>(m_total_samples) / static_cast<double>(m_sample_rate), checksum_string.to_string(), m_data_start_location, total_meta_blocks, total_meta_blocks - meta_blocks_parsed);
     }
 
     return true;
@@ -192,7 +192,7 @@ FlacRawMetadataBlock FlacLoaderPlugin::next_meta_block(InputBitStream& bit_input
     }
     m_data_start_location += 1;
 
-    u32 block_length = bit_input.read_bits_big_endian(24);
+    u32 block_length = static_cast<u32>(bit_input.read_bits_big_endian(24));
     m_data_start_location += 3;
     CHECK_IO_ERROR();
     auto block_data_result = ByteBuffer::create_uninitialized(block_length);
@@ -232,7 +232,7 @@ void FlacLoaderPlugin::seek(const int position)
 RefPtr<Buffer> FlacLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_input)
 {
     Vector<Sample> samples;
-    ssize_t remaining_samples = m_total_samples - m_loaded_samples;
+    ssize_t remaining_samples = static_cast<ssize_t>(m_total_samples - m_loaded_samples);
     if (remaining_samples <= 0) {
         return nullptr;
     }
@@ -247,7 +247,7 @@ RefPtr<Buffer> FlacLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_
             }
         }
         samples.append(m_current_frame_data.take_first());
-        if (m_current_frame_data.size() == 0) {
+        if (m_current_frame_data.is_empty()) {
             m_current_frame.clear();
         }
         --samples_to_read;
@@ -283,7 +283,7 @@ void FlacLoaderPlugin::next_frame()
     // TODO: Check the CRC-16 checksum (and others) by keeping track of read data
 
     // FLAC frame sync code starts header
-    u16 sync_code = bit_stream.read_bits_big_endian(14);
+    u16 sync_code = static_cast<u16>(bit_stream.read_bits_big_endian(14));
     ok = ok && (sync_code == 0b11111111111110);
     CHECK_OK("Sync code");
     bool reserved_bit = bit_stream.read_bit_big_endian();
@@ -291,20 +291,20 @@ void FlacLoaderPlugin::next_frame()
     CHECK_OK("Reserved frame header bit");
     [[maybe_unused]] bool blocking_strategy = bit_stream.read_bit_big_endian();
 
-    u32 sample_count = convert_sample_count_code(bit_stream.read_bits_big_endian(4));
+    u32 sample_count = convert_sample_count_code(static_cast<u8>(bit_stream.read_bits_big_endian(4)));
     CHECK_ERROR_STRING;
 
-    u32 frame_sample_rate = convert_sample_rate_code(bit_stream.read_bits_big_endian(4));
+    u32 frame_sample_rate = convert_sample_rate_code(static_cast<u8>(bit_stream.read_bits_big_endian(4)));
     CHECK_ERROR_STRING;
 
-    u8 channel_type_num = bit_stream.read_bits_big_endian(4);
+    u8 channel_type_num = static_cast<u8>(bit_stream.read_bits_big_endian(4));
     if (channel_type_num >= 0b1011) {
         ok = false;
         CHECK_OK("Channel assignment");
     }
     FlacFrameChannelType channel_type = (FlacFrameChannelType)channel_type_num;
 
-    PcmSampleFormat bit_depth = convert_bit_depth_code(bit_stream.read_bits_big_endian(3));
+    PcmSampleFormat bit_depth = convert_bit_depth_code(static_cast<u8>(bit_stream.read_bits_big_endian(3)));
     CHECK_ERROR_STRING;
 
     reserved_bit = bit_stream.read_bit_big_endian();
@@ -316,21 +316,21 @@ void FlacLoaderPlugin::next_frame()
 
     // Conditional header variables
     if (sample_count == FLAC_BLOCKSIZE_AT_END_OF_HEADER_8) {
-        sample_count = bit_stream.read_bits_big_endian(8) + 1;
+        sample_count = static_cast<u32>(bit_stream.read_bits_big_endian(8)) + 1;
     } else if (sample_count == FLAC_BLOCKSIZE_AT_END_OF_HEADER_16) {
-        sample_count = bit_stream.read_bits_big_endian(16) + 1;
+        sample_count = static_cast<u32>(bit_stream.read_bits_big_endian(16)) + 1;
     }
 
     if (frame_sample_rate == FLAC_SAMPLERATE_AT_END_OF_HEADER_8) {
-        frame_sample_rate = bit_stream.read_bits_big_endian(8) * 1000;
+        frame_sample_rate = static_cast<u32>(bit_stream.read_bits_big_endian(8)) * 1000;
     } else if (frame_sample_rate == FLAC_SAMPLERATE_AT_END_OF_HEADER_16) {
-        frame_sample_rate = bit_stream.read_bits_big_endian(16);
+        frame_sample_rate = static_cast<u32>(bit_stream.read_bits_big_endian(16));
     } else if (frame_sample_rate == FLAC_SAMPLERATE_AT_END_OF_HEADER_16X10) {
-        frame_sample_rate = bit_stream.read_bits_big_endian(16) * 10;
+        frame_sample_rate = static_cast<u32>(bit_stream.read_bits_big_endian(16)) * 10;
     }
 
     // TODO: check header checksum, see above
-    [[maybe_unused]] u8 checksum = bit_stream.read_bits(8);
+    [[maybe_unused]] u8 checksum = static_cast<u8>(bit_stream.read_bits(8));
 
     dbgln_if(AFLACLOADER_DEBUG, "Frame: {} samples, {}bit {}Hz, channeltype {:x}, {} number {}, header checksum {}", sample_count, pcm_bits_per_sample(bit_depth), frame_sample_rate, channel_type_num, blocking_strategy ? "sample" : "frame", m_current_sample_or_frame, checksum);
 
@@ -356,9 +356,10 @@ void FlacLoaderPlugin::next_frame()
     bit_stream.align_to_byte_boundary();
 
     // TODO: check checksum, see above
-    [[maybe_unused]] u16 footer_checksum = bit_stream.read_bits_big_endian(16);
+    [[maybe_unused]] u16 footer_checksum = static_cast<u16>(bit_stream.read_bits_big_endian(16));
 
-    Vector<i32> left, right;
+    Vector<i32> left;
+    Vector<i32> right;
 
     switch (channel_type) {
     case FlacFrameChannelType::Mono:
@@ -514,7 +515,7 @@ u8 frame_channel_type_to_channel_count(FlacFrameChannelType channel_type)
 
 FlacSubframeHeader FlacLoaderPlugin::next_subframe_header(InputBitStream& bit_stream, u8 channel_index)
 {
-    u8 bits_per_sample = pcm_bits_per_sample(m_current_frame->bit_depth);
+    u8 bits_per_sample = static_cast<u16>(pcm_bits_per_sample(m_current_frame->bit_depth));
 
     // For inter-channel correlation, the side channel needs an extra bit for its samples
     switch (m_current_frame->channels) {
@@ -541,7 +542,7 @@ FlacSubframeHeader FlacLoaderPlugin::next_subframe_header(InputBitStream& bit_st
     };
 
     // subframe type (encoding)
-    u8 subframe_code = bit_stream.read_bits_big_endian(6);
+    u8 subframe_code = static_cast<u8>(bit_stream.read_bits_big_endian(6));
     if ((subframe_code >= 0b000010 && subframe_code <= 0b000111) || (subframe_code > 0b001100 && subframe_code < 0b100000)) {
         m_error_string = "Subframe type";
         return {};
@@ -590,8 +591,10 @@ Vector<i32> FlacLoaderPlugin::parse_subframe(FlacSubframeHeader& subframe_header
         dbgln_if(AFLACLOADER_DEBUG, "Constant subframe: {}", constant_value);
 
         samples.ensure_capacity(m_current_frame->sample_count);
+        VERIFY(subframe_header.bits_per_sample - subframe_header.wasted_bits_per_sample != 0);
+        i32 constant = sign_extend(static_cast<u32>(constant_value), subframe_header.bits_per_sample - subframe_header.wasted_bits_per_sample);
         for (u32 i = 0; i < m_current_frame->sample_count; ++i) {
-            samples.unchecked_append(sign_extend(constant_value, subframe_header.bits_per_sample - subframe_header.wasted_bits_per_sample));
+            samples.unchecked_append(constant);
         }
         break;
     }
@@ -632,8 +635,11 @@ Vector<i32> FlacLoaderPlugin::decode_verbatim(FlacSubframeHeader& subframe, Inpu
     Vector<i32> decoded;
     decoded.ensure_capacity(m_current_frame->sample_count);
 
+    VERIFY(subframe.bits_per_sample - subframe.wasted_bits_per_sample != 0);
     for (size_t i = 0; i < m_current_frame->sample_count; ++i) {
-        decoded.unchecked_append(sign_extend(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample), subframe.bits_per_sample - subframe.wasted_bits_per_sample));
+        decoded.unchecked_append(sign_extend(
+            static_cast<u32>(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample)),
+            subframe.bits_per_sample - subframe.wasted_bits_per_sample));
     }
 
     return decoded;
@@ -645,13 +651,16 @@ Vector<i32> FlacLoaderPlugin::decode_custom_lpc(FlacSubframeHeader& subframe, In
     Vector<i32> decoded;
     decoded.ensure_capacity(m_current_frame->sample_count);
 
+    VERIFY(subframe.bits_per_sample - subframe.wasted_bits_per_sample != 0);
     // warm-up samples
     for (auto i = 0; i < subframe.order; ++i) {
-        decoded.unchecked_append(sign_extend(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample), subframe.bits_per_sample - subframe.wasted_bits_per_sample));
+        decoded.unchecked_append(sign_extend(
+            static_cast<u32>(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample)),
+            subframe.bits_per_sample - subframe.wasted_bits_per_sample));
     }
 
     // precision of the coefficients
-    u8 lpc_precision = bit_input.read_bits_big_endian(4);
+    u8 lpc_precision = static_cast<u8>(bit_input.read_bits_big_endian(4));
     if (lpc_precision == 0b1111) {
         m_error_string = "Invalid linear predictor coefficient precision";
         return {};
@@ -659,14 +668,14 @@ Vector<i32> FlacLoaderPlugin::decode_custom_lpc(FlacSubframeHeader& subframe, In
     lpc_precision += 1;
 
     // shift needed on the data (signed!)
-    i8 lpc_shift = sign_extend(bit_input.read_bits_big_endian(5), 5);
+    i8 lpc_shift = sign_extend(static_cast<u32>(bit_input.read_bits_big_endian(5)), 5);
 
     Vector<i32> coefficients;
     coefficients.ensure_capacity(subframe.order);
     // read coefficients
     for (auto i = 0; i < subframe.order; ++i) {
-        u32 raw_coefficient = bit_input.read_bits_big_endian(lpc_precision);
-        i32 coefficient = sign_extend(raw_coefficient, lpc_precision);
+        u32 raw_coefficient = static_cast<u32>(bit_input.read_bits_big_endian(lpc_precision));
+        i32 coefficient = static_cast<i32>(sign_extend(raw_coefficient, lpc_precision));
         coefficients.unchecked_append(coefficient);
     }
 
@@ -694,9 +703,12 @@ Vector<i32> FlacLoaderPlugin::decode_fixed_lpc(FlacSubframeHeader& subframe, Inp
     Vector<i32> decoded;
     decoded.ensure_capacity(m_current_frame->sample_count);
 
+    VERIFY(subframe.bits_per_sample - subframe.wasted_bits_per_sample != 0);
     // warm-up samples
     for (auto i = 0; i < subframe.order; ++i) {
-        decoded.unchecked_append(sign_extend(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample), subframe.bits_per_sample - subframe.wasted_bits_per_sample));
+        decoded.unchecked_append(sign_extend(
+            static_cast<u32>(bit_input.read_bits_big_endian(subframe.bits_per_sample - subframe.wasted_bits_per_sample)),
+            subframe.bits_per_sample - subframe.wasted_bits_per_sample));
     }
 
     decode_residual(decoded, subframe, bit_input);
@@ -740,8 +752,8 @@ Vector<i32> FlacLoaderPlugin::decode_fixed_lpc(FlacSubframeHeader& subframe, Inp
 // Decode the residual, the "error" between the function approximation and the actual audio data
 Vector<i32> FlacLoaderPlugin::decode_residual(Vector<i32>& decoded, FlacSubframeHeader& subframe, InputBitStream& bit_input)
 {
-    u8 residual_mode = bit_input.read_bits_big_endian(2);
-    u8 partition_order = bit_input.read_bits_big_endian(4);
+    u8 residual_mode = static_cast<u8>(bit_input.read_bits_big_endian(2));
+    u8 partition_order = static_cast<u8>(bit_input.read_bits_big_endian(4));
     size_t partitions = 1 << partition_order;
 
     if (residual_mode == FlacResidualMode::Rice4Bit) {
@@ -768,7 +780,7 @@ Vector<i32> FlacLoaderPlugin::decode_residual(Vector<i32>& decoded, FlacSubframe
 ALWAYS_INLINE Vector<i32> FlacLoaderPlugin::decode_rice_partition(u8 partition_type, u32 partitions, u32 partition_index, FlacSubframeHeader& subframe, InputBitStream& bit_input)
 {
     // Rice parameter / Exp-Golomb order
-    u8 k = bit_input.read_bits_big_endian(partition_type);
+    u8 k = static_cast<u8>(bit_input.read_bits_big_endian(partition_type));
 
     u32 residual_sample_count;
     if (partitions == 0)
@@ -783,9 +795,9 @@ ALWAYS_INLINE Vector<i32> FlacLoaderPlugin::decode_rice_partition(u8 partition_t
 
     // escape code for unencoded binary partition
     if (k == (1 << partition_type) - 1) {
-        u8 unencoded_bps = bit_input.read_bits_big_endian(5);
+        u8 unencoded_bps = static_cast<u8>(bit_input.read_bits_big_endian(5));
         for (size_t r = 0; r < residual_sample_count; ++r) {
-            rice_partition[r] = bit_input.read_bits_big_endian(unencoded_bps);
+            rice_partition[r] = static_cast<u8>(bit_input.read_bits_big_endian(unencoded_bps));
         }
     } else {
         for (size_t r = 0; r < residual_sample_count; ++r) {
@@ -804,8 +816,8 @@ ALWAYS_INLINE i32 decode_unsigned_exp_golomb(u8 k, InputBitStream& bit_input)
         ++q;
 
     // least significant bits (remainder)
-    u32 rem = bit_input.read_bits_big_endian(k);
-    u32 value = (u32)(q << k | rem);
+    u32 rem = static_cast<u32>(bit_input.read_bits_big_endian(k));
+    u32 value = q << k | rem;
 
     return rice_to_signed(value);
 }
@@ -853,8 +865,8 @@ i32 rice_to_signed(u32 x)
 {
     // positive numbers are even, negative numbers are odd
     // bitmask for conditionally inverting the entire number, thereby "negating" it
-    i32 sign = -(x & 1);
+    i32 sign = -static_cast<i32>(x & 1);
     // copies the sign's sign onto the actual magnitude of x
-    return (i32)(sign ^ (x >> 1));
+    return static_cast<i32>(sign ^ (x >> 1));
 }
 }
diff --git a/Userland/Libraries/LibAudio/FlacLoader.h b/Userland/Libraries/LibAudio/FlacLoader.h
index 9464204d8a..dd7da0e423 100644
--- a/Userland/Libraries/LibAudio/FlacLoader.h
+++ b/Userland/Libraries/LibAudio/FlacLoader.h
@@ -69,8 +69,8 @@ ALWAYS_INLINE i32 decode_unsigned_exp_golomb(u8 order, InputBitStream& bit_input
 
 class FlacLoaderPlugin : public LoaderPlugin {
 public:
-    FlacLoaderPlugin(StringView path);
-    FlacLoaderPlugin(const ByteBuffer& buffer);
+    explicit FlacLoaderPlugin(StringView path);
+    explicit FlacLoaderPlugin(const ByteBuffer& buffer);
     ~FlacLoaderPlugin()
     {
         if (m_stream)
@@ -87,8 +87,8 @@ public:
     virtual void reset() override;
     virtual void seek(const int position) override;
 
-    virtual int loaded_samples() override { return m_loaded_samples; }
-    virtual int total_samples() override { return m_total_samples; }
+    virtual int loaded_samples() override { return static_cast<int>(m_loaded_samples); }
+    virtual int total_samples() override { return static_cast<int>(m_total_samples); }
     virtual u32 sample_rate() override { return m_sample_rate; }
     virtual u16 num_channels() override { return m_num_channels; }
     virtual PcmSampleFormat pcm_format() override { return m_sample_format; }
diff --git a/Userland/Libraries/LibAudio/Loader.h b/Userland/Libraries/LibAudio/Loader.h
index 11d284e1df..276811e5a6 100644
--- a/Userland/Libraries/LibAudio/Loader.h
+++ b/Userland/Libraries/LibAudio/Loader.h
@@ -78,8 +78,8 @@ public:
     RefPtr<Core::File> file() const { return m_plugin ? m_plugin->file() : nullptr; }
 
 private:
-    Loader(StringView path);
-    Loader(const ByteBuffer& buffer);
+    explicit Loader(StringView path);
+    explicit Loader(const ByteBuffer& buffer);
 
     mutable OwnPtr<LoaderPlugin> m_plugin;
 };
diff --git a/Userland/Libraries/LibAudio/Sample.h b/Userland/Libraries/LibAudio/Sample.h
index 50e3c84907..e6b00b8456 100644
--- a/Userland/Libraries/LibAudio/Sample.h
+++ b/Userland/Libraries/LibAudio/Sample.h
@@ -10,7 +10,8 @@
 #include <AK/Math.h>
 
 namespace Audio {
-using namespace AK::Exponentials;
+using AK::Exponentials::exp;
+using AK::Exponentials::log;
 // Constants for logarithmic volume. See Sample::linear_to_log
 // Corresponds to 60dB
 constexpr double DYNAMIC_RANGE = 1000;
@@ -23,7 +24,7 @@ struct Sample {
     constexpr Sample() = default;
 
     // For mono
-    constexpr Sample(double left)
+    constexpr explicit Sample(double left)
         : left(left)
         , right(left)
     {
@@ -63,13 +64,13 @@ struct Sample {
     // - Linear:        0.0 to 1.0
     // - Logarithmic:   0.0 to 1.0
 
-    ALWAYS_INLINE double linear_to_log(double const change)
+    ALWAYS_INLINE double linear_to_log(double const change) const
     {
         // TODO: Add linear slope around 0
         return VOLUME_A * exp(VOLUME_B * change);
     }
 
-    ALWAYS_INLINE double log_to_linear(double const val)
+    ALWAYS_INLINE double log_to_linear(double const val) const
     {
         // TODO: Add linear slope around 0
         return log(val / VOLUME_A) / VOLUME_B;
diff --git a/Userland/Libraries/LibAudio/WavLoader.h b/Userland/Libraries/LibAudio/WavLoader.h
index 1ae39dffb4..2a2976536b 100644
--- a/Userland/Libraries/LibAudio/WavLoader.h
+++ b/Userland/Libraries/LibAudio/WavLoader.h
@@ -33,8 +33,8 @@ class Buffer;
 // Parses a WAV file and produces an Audio::Buffer.
 class WavLoaderPlugin : public LoaderPlugin {
 public:
-    WavLoaderPlugin(StringView path);
-    WavLoaderPlugin(const ByteBuffer& buffer);
+    explicit WavLoaderPlugin(StringView path);
+    explicit WavLoaderPlugin(const ByteBuffer& buffer);
 
     virtual bool sniff() override { return valid; }
 
diff --git a/Userland/Libraries/LibAudio/WavWriter.cpp b/Userland/Libraries/LibAudio/WavWriter.cpp
index b4d7f0873e..4428a38f9d 100644
--- a/Userland/Libraries/LibAudio/WavWriter.cpp
+++ b/Userland/Libraries/LibAudio/WavWriter.cpp
@@ -8,7 +8,7 @@
 
 namespace Audio {
 
-WavWriter::WavWriter(StringView path, int sample_rate, int num_channels, int bits_per_sample)
+WavWriter::WavWriter(StringView path, int sample_rate, u16 num_channels, u16 bits_per_sample)
     : m_sample_rate(sample_rate)
     , m_num_channels(num_channels)
     , m_bits_per_sample(bits_per_sample)
@@ -16,7 +16,7 @@ WavWriter::WavWriter(StringView path, int sample_rate, int num_channels, int bit
     set_file(path);
 }
 
-WavWriter::WavWriter(int sample_rate, int num_channels, int bits_per_sample)
+WavWriter::WavWriter(int sample_rate, u16 num_channels, u16 bits_per_sample)
     : m_sample_rate(sample_rate)
     , m_num_channels(num_channels)
     , m_bits_per_sample(bits_per_sample)
diff --git a/Userland/Libraries/LibAudio/WavWriter.h b/Userland/Libraries/LibAudio/WavWriter.h
index 555bc09cc2..d36059ab12 100644
--- a/Userland/Libraries/LibAudio/WavWriter.h
+++ b/Userland/Libraries/LibAudio/WavWriter.h
@@ -6,15 +6,19 @@
 
 #pragma once
 
+#include <AK/Noncopyable.h>
 #include <AK/StringView.h>
 #include <LibCore/File.h>
 
 namespace Audio {
 
 class WavWriter {
+    AK_MAKE_NONCOPYABLE(WavWriter);
+    AK_MAKE_NONMOVABLE(WavWriter);
+
 public:
-    WavWriter(StringView path, int sample_rate = 44100, int num_channels = 2, int bits_per_sample = 16);
-    WavWriter(int sample_rate = 44100, int num_channels = 2, int bits_per_sample = 16);
+    WavWriter(StringView path, int sample_rate = 44100, u16 num_channels = 2, u16 bits_per_sample = 16);
+    WavWriter(int sample_rate = 44100, u16 num_channels = 2, u16 bits_per_sample = 16);
     ~WavWriter();
 
     bool has_error() const { return !m_error_string.is_null(); }
diff --git a/Userland/Libraries/LibDSP/Music.h b/Userland/Libraries/LibDSP/Music.h
index 30059f22b1..fed5112571 100644
--- a/Userland/Libraries/LibDSP/Music.h
+++ b/Userland/Libraries/LibDSP/Music.h
@@ -40,8 +40,11 @@ struct Signal : public Variant<Sample, RollNotes> {
     using Variant::Variant;
     ALWAYS_INLINE SignalType type() const
     {
-        return has<Sample>() ? SignalType::Sample : has<RollNotes>() ? SignalType::Note
-                                                                     : SignalType::Invalid;
+        if (has<Sample>())
+            return SignalType::Sample;
+        if (has<RollNotes>())
+            return SignalType::Note;
+        return SignalType::Invalid;
     }
 };
 
diff --git a/Userland/Libraries/LibDSP/Track.h b/Userland/Libraries/LibDSP/Track.h
index af873f791c..0b902cff89 100644
--- a/Userland/Libraries/LibDSP/Track.h
+++ b/Userland/Libraries/LibDSP/Track.h
@@ -19,7 +19,6 @@ class Track : public Core::Object {
 public:
     Track(NonnullRefPtr<Transport> transport)
         : m_transport(move(transport))
-        , m_current_signal(Sample {})
     {
     }
     virtual ~Track() override = default;
@@ -42,7 +41,7 @@ protected:
     NonnullRefPtrVector<Processor> m_processor_chain;
     NonnullRefPtr<Transport> m_transport;
     // The current signal is stored here, to prevent unnecessary reallocation.
-    Signal m_current_signal;
+    Signal m_current_signal { Audio::Sample {} };
 };
 
 class NoteTrack final : public Track {
@@ -53,7 +52,7 @@ public:
     NonnullRefPtrVector<NoteClip> const& clips() const { return m_clips; }
 
 protected:
-    virtual void compute_current_clips_signal() override;
+    void compute_current_clips_signal() override;
 
 private:
     NonnullRefPtrVector<NoteClip> m_clips;
@@ -67,7 +66,7 @@ public:
     NonnullRefPtrVector<AudioClip> const& clips() const { return m_clips; }
 
 protected:
-    virtual void compute_current_clips_signal() override;
+    void compute_current_clips_signal() override;
 
 private:
     NonnullRefPtrVector<AudioClip> m_clips;
diff --git a/Userland/Services/AudioServer/Mixer.cpp b/Userland/Services/AudioServer/Mixer.cpp
index 99d6038930..52a5761735 100644
--- a/Userland/Services/AudioServer/Mixer.cpp
+++ b/Userland/Services/AudioServer/Mixer.cpp
@@ -114,7 +114,7 @@ void Mixer::mix()
 
                 // Even though it's not realistic, the user expects no sound at 0%.
                 if (m_main_volume < 0.01)
-                    mixed_sample = { 0 };
+                    mixed_sample = Audio::Sample { 0 };
                 else
                     mixed_sample.log_multiply(m_main_volume);
                 mixed_sample.clip();