LibAudio: Support 32 and 64-bit float WAV files

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)

5 files changed, 127 insertions, 35 deletions

diff --git a/Userland/Libraries/LibAudio/Buffer.cpp b/Userland/Libraries/LibAudio/Buffer.cpp
index 2e6ae9d24c..ce3da09cd8 100644
--- a/Userland/Libraries/LibAudio/Buffer.cpp
+++ b/Userland/Libraries/LibAudio/Buffer.cpp
@@ -1,14 +1,41 @@
 /*
  * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
+ * Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
  *
  * SPDX-License-Identifier: BSD-2-Clause
  */
 
 #include <AK/Atomic.h>
+#include <AK/Debug.h>
+#include <AK/String.h>
 #include <LibAudio/Buffer.h>
 
 namespace Audio {
 
+u16 pcm_bits_per_sample(PcmSampleFormat format)
+{
+    switch (format) {
+    case Uint8:
+        return 8;
+    case Int16:
+        return 16;
+    case Int24:
+        return 24;
+    case Float32:
+        return 32;
+    case Float64:
+        return 64;
+    default:
+        VERIFY_NOT_REACHED();
+    }
+}
+
+String sample_format_name(PcmSampleFormat format)
+{
+    bool is_float = format == Float32 || format == Float64;
+    return String::formatted("PCM {}bit {}", pcm_bits_per_sample(format), is_float ? "Float" : "LE");
+}
+
 i32 Buffer::allocate_id()
 {
     static Atomic<i32> next_id;
@@ -54,6 +81,20 @@ static void read_samples_from_stream(InputMemoryStream& stream, SampleReader rea
     }
 }
 
+static double read_float_sample_64(InputMemoryStream& stream)
+{
+    LittleEndian<double> sample;
+    stream >> sample;
+    return double(sample);
+}
+
+static double read_float_sample_32(InputMemoryStream& stream)
+{
+    LittleEndian<float> sample;
+    stream >> sample;
+    return double(sample);
+}
+
 static double read_norm_sample_24(InputMemoryStream& stream)
 {
     u8 byte = 0;
@@ -85,27 +126,33 @@ static double read_norm_sample_8(InputMemoryStream& stream)
     return double(sample) / NumericLimits<u8>::max();
 }
 
-RefPtr<Buffer> Buffer::from_pcm_data(ReadonlyBytes data, ResampleHelper& resampler, int num_channels, int bits_per_sample)
+RefPtr<Buffer> Buffer::from_pcm_data(ReadonlyBytes data, ResampleHelper& resampler, int num_channels, PcmSampleFormat sample_format)
 {
     InputMemoryStream stream { data };
-    return from_pcm_stream(stream, resampler, num_channels, bits_per_sample, data.size() / (bits_per_sample / 8));
+    return from_pcm_stream(stream, resampler, num_channels, sample_format, data.size() / (pcm_bits_per_sample(sample_format) / 8));
 }
 
-RefPtr<Buffer> Buffer::from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, int bits_per_sample, int num_samples)
+RefPtr<Buffer> Buffer::from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, PcmSampleFormat sample_format, int num_samples)
 {
     Vector<Frame> fdata;
     fdata.ensure_capacity(num_samples);
 
-    switch (bits_per_sample) {
-    case 8:
+    switch (sample_format) {
+    case PcmSampleFormat::Uint8:
         read_samples_from_stream(stream, read_norm_sample_8, fdata, resampler, num_channels);
         break;
-    case 16:
+    case PcmSampleFormat::Int16:
         read_samples_from_stream(stream, read_norm_sample_16, fdata, resampler, num_channels);
         break;
-    case 24:
+    case PcmSampleFormat::Int24:
         read_samples_from_stream(stream, read_norm_sample_24, fdata, resampler, num_channels);
         break;
+    case PcmSampleFormat::Float32:
+        read_samples_from_stream(stream, read_float_sample_32, fdata, resampler, num_channels);
+        break;
+    case PcmSampleFormat::Float64:
+        read_samples_from_stream(stream, read_float_sample_64, fdata, resampler, num_channels);
+        break;
     default:
         VERIFY_NOT_REACHED();
     }
diff --git a/Userland/Libraries/LibAudio/Buffer.h b/Userland/Libraries/LibAudio/Buffer.h
index fda5e0ef9c..bcdb9a8279 100644
--- a/Userland/Libraries/LibAudio/Buffer.h
+++ b/Userland/Libraries/LibAudio/Buffer.h
@@ -1,5 +1,6 @@
 /*
  * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
+ * Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
  *
  * SPDX-License-Identifier: BSD-2-Clause
  */
@@ -8,6 +9,7 @@
 
 #include <AK/ByteBuffer.h>
 #include <AK/MemoryStream.h>
+#include <AK/String.h>
 #include <AK/Types.h>
 #include <AK/Vector.h>
 #include <LibCore/AnonymousBuffer.h>
@@ -69,6 +71,19 @@ struct Frame {
     double right;
 };
 
+// Supported PCM sample formats.
+enum PcmSampleFormat : u8 {
+    Uint8,
+    Int16,
+    Int24,
+    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...
@@ -89,8 +104,8 @@ private:
 // A buffer of audio samples, normalized to 44100hz.
 class Buffer : public RefCounted<Buffer> {
 public:
-    static RefPtr<Buffer> from_pcm_data(ReadonlyBytes data, ResampleHelper& resampler, int num_channels, int bits_per_sample);
-    static RefPtr<Buffer> from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, int num_channels, int bits_per_sample, int num_samples);
+    static RefPtr<Buffer> from_pcm_data(ReadonlyBytes data, ResampleHelper& resampler, int num_channels, PcmSampleFormat sample_format);
+    static RefPtr<Buffer> from_pcm_stream(InputMemoryStream& stream, ResampleHelper& resampler, 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)));
diff --git a/Userland/Libraries/LibAudio/Loader.h b/Userland/Libraries/LibAudio/Loader.h
index 5cd605934d..0050d6acd3 100644
--- a/Userland/Libraries/LibAudio/Loader.h
+++ b/Userland/Libraries/LibAudio/Loader.h
@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2018-2020, the SerenityOS developers.
+ * Copyright (c) 2018-2021, the SerenityOS developers.
  *
  * SPDX-License-Identifier: BSD-2-Clause
  */
@@ -33,7 +33,7 @@ public:
     virtual int total_samples() = 0;
     virtual u32 sample_rate() = 0;
     virtual u16 num_channels() = 0;
-    virtual u16 bits_per_sample() = 0;
+    virtual PcmSampleFormat pcm_format() = 0;
     virtual RefPtr<Core::File> file() = 0;
 };
 
@@ -62,7 +62,7 @@ public:
     int total_samples() const { return m_plugin ? m_plugin->total_samples() : 0; }
     u32 sample_rate() const { return m_plugin ? m_plugin->sample_rate() : 0; }
     u16 num_channels() const { return m_plugin ? m_plugin->num_channels() : 0; }
-    u16 bits_per_sample() const { return m_plugin ? m_plugin->bits_per_sample() : 0; }
+    u16 bits_per_sample() const { return m_plugin ? pcm_bits_per_sample(m_plugin->pcm_format()) : 0; }
     RefPtr<Core::File> file() const { return m_plugin ? m_plugin->file() : nullptr; }
 
 private:
diff --git a/Userland/Libraries/LibAudio/WavLoader.cpp b/Userland/Libraries/LibAudio/WavLoader.cpp
index 508a33e12d..b300256e42 100644
--- a/Userland/Libraries/LibAudio/WavLoader.cpp
+++ b/Userland/Libraries/LibAudio/WavLoader.cpp
@@ -1,5 +1,6 @@
 /*
  * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
+ * Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
  *
  * SPDX-License-Identifier: BSD-2-Clause
  */
@@ -54,17 +55,18 @@ bool WavLoaderPlugin::sniff()
 RefPtr<Buffer> WavLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_input)
 {
 #if AWAVLOADER_DEBUG
-    dbgln("Read WAV of format PCM with num_channels {} sample rate {}, bits per sample {}", m_num_channels, m_sample_rate, m_bits_per_sample);
+    dbgln("Read {} bytes WAV with num_channels {} sample rate {}, bits per sample {}, sample format {}", max_bytes_to_read_from_input, m_num_channels, m_sample_rate, pcm_bits_per_sample(m_sample_format), sample_format_name(m_sample_format));
 #endif
-    size_t samples_to_read = static_cast<int>(max_bytes_to_read_from_input) / (m_num_channels * (m_bits_per_sample / 8));
+    size_t samples_to_read = static_cast<int>(max_bytes_to_read_from_input) / (m_num_channels * (pcm_bits_per_sample(m_sample_format) / 8));
     RefPtr<Buffer> buffer;
     if (m_file) {
         auto raw_samples = m_file->read(max_bytes_to_read_from_input);
-        if (raw_samples.is_empty())
+        if (raw_samples.is_empty()) {
             return nullptr;
-        buffer = Buffer::from_pcm_data(raw_samples, *m_resampler, m_num_channels, m_bits_per_sample);
+        }
+        buffer = Buffer::from_pcm_data(raw_samples, *m_resampler, m_num_channels, m_sample_format);
     } else {
-        buffer = Buffer::from_pcm_stream(*m_stream, *m_resampler, m_num_channels, m_bits_per_sample, samples_to_read);
+        buffer = Buffer::from_pcm_stream(*m_stream, *m_resampler, m_num_channels, m_sample_format, samples_to_read);
     }
     //Buffer contains normalized samples, but m_loaded_samples should contain the amount of actually loaded samples
     m_loaded_samples += samples_to_read;
@@ -78,7 +80,7 @@ void WavLoaderPlugin::seek(const int position)
         return;
 
     m_loaded_samples = position;
-    size_t byte_position = position * m_num_channels * (m_bits_per_sample / 8);
+    size_t byte_position = position * m_num_channels * (pcm_bits_per_sample(m_sample_format) / 8);
 
     if (m_file)
         m_file->seek(byte_position);
@@ -147,7 +149,7 @@ bool WavLoaderPlugin::parse_header()
             m_error_string = String::formatted("Parsing failed: {}", msg); \
             return {};                                                     \
         }                                                                  \
-    } while (0);
+    } while (0)
 
     u32 riff = read_u32();
     ok = ok && riff == 0x46464952; // "RIFF"
@@ -156,7 +158,6 @@ bool WavLoaderPlugin::parse_header()
     u32 sz = read_u32();
     ok = ok && sz < 1024 * 1024 * 1024; // arbitrary
     CHECK_OK("File size");
-    VERIFY(sz < 1024 * 1024 * 1024);
 
     u32 wave = read_u32();
     ok = ok && wave == 0x45564157; // "WAVE"
@@ -169,13 +170,11 @@ bool WavLoaderPlugin::parse_header()
     u32 fmt_size = read_u32();
     ok = ok && fmt_size == 16;
     CHECK_OK("FMT size");
-    VERIFY(fmt_size == 16);
 
     u16 audio_format = read_u16();
-    CHECK_OK("Audio format");     // incomplete read check
-    ok = ok && audio_format == 1; // WAVE_FORMAT_PCM
-    CHECK_OK("Audio format");     // value check
-    VERIFY(audio_format == 1);
+    CHECK_OK("Audio format"); // incomplete read check
+    ok = ok && (audio_format == WAVE_FORMAT_PCM || audio_format == WAVE_FORMAT_IEEE_FLOAT);
+    CHECK_OK("Audio format PCM/Float"); // value check
 
     m_num_channels = read_u16();
     ok = ok && (m_num_channels == 1 || m_num_channels == 2);
@@ -185,16 +184,40 @@ bool WavLoaderPlugin::parse_header()
     CHECK_OK("Sample rate");
 
     read_u32();
-    CHECK_OK("Byte rate");
+    CHECK_OK("Data rate");
 
     read_u16();
-    CHECK_OK("Block align");
+    CHECK_OK("Block size");
 
-    m_bits_per_sample = read_u16();
+    u16 bits_per_sample = read_u16();
     CHECK_OK("Bits per sample"); // incomplete read check
-    ok = ok && (m_bits_per_sample == 8 || m_bits_per_sample == 16 || m_bits_per_sample == 24);
-    CHECK_OK("Bits per sample"); // value check
-    VERIFY(m_bits_per_sample == 8 || m_bits_per_sample == 16 || m_bits_per_sample == 24);
+    if (audio_format == WAVE_FORMAT_PCM) {
+        ok = ok && (bits_per_sample == 8 || bits_per_sample == 16 || bits_per_sample == 24);
+        CHECK_OK("Bits per sample (PCM)"); // value check
+
+        // We only support 8-24 bit audio right now because other formats are uncommon
+        if (bits_per_sample == 8) {
+            m_sample_format = PcmSampleFormat::Uint8;
+        } else if (bits_per_sample == 16) {
+            m_sample_format = PcmSampleFormat::Int16;
+        } else if (bits_per_sample == 24) {
+            m_sample_format = PcmSampleFormat::Int24;
+        }
+    } else if (audio_format == WAVE_FORMAT_IEEE_FLOAT) {
+        ok = ok && (bits_per_sample == 32 || bits_per_sample == 64);
+        CHECK_OK("Bits per sample (Float)"); // value check
+
+        // Again, only the common 32 and 64 bit
+        if (bits_per_sample == 32) {
+            m_sample_format = PcmSampleFormat::Float32;
+        } else if (bits_per_sample == 64) {
+            m_sample_format = PcmSampleFormat::Float64;
+        }
+    }
+
+#if AWAVLOADER_DEBUG
+    dbgln("WAV format {} at {}bit, {} channels, rate {}Hz ", sample_format_name(m_sample_format), pcm_bits_per_sample(m_sample_format), m_num_channels, m_sample_rate);
+#endif
 
     // Read chunks until we find DATA
     bool found_data = false;
@@ -223,12 +246,11 @@ bool WavLoaderPlugin::parse_header()
 
     ok = ok && found_data;
     CHECK_OK("Found no data chunk");
-    VERIFY(found_data);
 
     ok = ok && data_sz < maximum_wav_size;
     CHECK_OK("Data was too large");
 
-    int bytes_per_sample = (m_bits_per_sample / 8) * m_num_channels;
+    int bytes_per_sample = (bits_per_sample / 8) * m_num_channels;
     m_total_samples = data_sz / bytes_per_sample;
 
     return true;
diff --git a/Userland/Libraries/LibAudio/WavLoader.h b/Userland/Libraries/LibAudio/WavLoader.h
index c8077dd15f..bab6ac00a4 100644
--- a/Userland/Libraries/LibAudio/WavLoader.h
+++ b/Userland/Libraries/LibAudio/WavLoader.h
@@ -1,5 +1,6 @@
 /*
  * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
+ * Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
  *
  * SPDX-License-Identifier: BSD-2-Clause
  */
@@ -19,6 +20,13 @@
 namespace Audio {
 class Buffer;
 
+// defines for handling the WAV header data
+#define WAVE_FORMAT_PCM 0x0001        // PCM
+#define WAVE_FORMAT_IEEE_FLOAT 0x0003 // IEEE float
+#define WAVE_FORMAT_ALAW 0x0006       // 8-bit ITU-T G.711 A-law
+#define WAVE_FORMAT_MULAW 0x0007      // 8-bit ITU-T G.711 µ-law
+#define WAVE_FORMAT_EXTENSIBLE 0xFFFE // Determined by SubFormat
+
 // Parses a WAV file and produces an Audio::Buffer.
 class WavLoaderPlugin : public LoaderPlugin {
 public:
@@ -39,7 +47,7 @@ public:
     virtual int total_samples() override { return m_total_samples; }
     virtual u32 sample_rate() override { return m_sample_rate; }
     virtual u16 num_channels() override { return m_num_channels; }
-    virtual u16 bits_per_sample() override { return m_bits_per_sample; }
+    virtual PcmSampleFormat pcm_format() override { return m_sample_format; }
     virtual RefPtr<Core::File> file() override { return m_file; }
 
 private:
@@ -53,7 +61,7 @@ private:
 
     u32 m_sample_rate { 0 };
     u16 m_num_channels { 0 };
-    u16 m_bits_per_sample { 0 };
+    PcmSampleFormat m_sample_format;
 
     int m_loaded_samples { 0 };
     int m_total_samples { 0 };