Audio: Make basic streaming WAV playback work.

I had to solve a bunch of things simultaneously to make this work.
Refactor AWavLoader to be a streaming loader rather than a one-shot one.
The constructor parses the header, and if everything looks good, you can
repeatedly ask the AWavLoader for sample buffers until it runs out.

Also send a message from AudioServer when a buffer has finished playing.
That allows us to implement a blocking variant of play().

Use all of this in aplay to play WAV files chunk-at-a-time.
This is definitely not perfect and it's a little glitchy and skippy,
but I think it's a step in the right direction.

Libraries/LibAudio/ABuffer.h

@@ -53,9 +53,10 @@ struct ASample {
 class ABuffer : public RefCounted<ABuffer> {
 public:
     static RefPtr<ABuffer> from_pcm_data(ByteBuffer& data, int num_channels, int bits_per_sample, int source_rate);
-    ABuffer(Vector<ASample>&& samples)
+    static NonnullRefPtr<ABuffer> create_with_samples(Vector<ASample>&& samples)
-        : m_samples(move(samples))
+    {
-    {}
+        return adopt(*new ABuffer(move(samples)));
+    }
 
     const Vector<ASample>& samples() const { return m_samples; }
     Vector<ASample>& samples() { return m_samples; }
@@ -63,6 +64,10 @@ public:
     int size_in_bytes() const { return m_samples.size() * sizeof(ASample); }
 
 private:
+    ABuffer(Vector<ASample>&& samples)
+        : m_samples(move(samples))
+    {
+    }
+
     Vector<ASample> m_samples;
 };
-

Libraries/LibAudio/AClientConnection.cpp

@@ -17,7 +17,7 @@ void AClientConnection::handshake()
     set_my_client_id(response.greeting.your_client_id);
 }
 
-void AClientConnection::play(const ABuffer& buffer)
+void AClientConnection::play(const ABuffer& buffer, bool block)
 {
     auto shared_buf = SharedBuffer::create_with_size(buffer.size_in_bytes());
     if (!shared_buf) {
@@ -31,5 +31,5 @@ void AClientConnection::play(const ABuffer& buffer)
     ASAPI_ClientMessage request;
     request.type = ASAPI_ClientMessage::Type::PlayBuffer;
     request.play_buffer.buffer_id = shared_buf->shared_buffer_id();
-    sync_request(request, ASAPI_ServerMessage::Type::PlayingBuffer);
+    sync_request(request, block ? ASAPI_ServerMessage::Type::FinishedPlayingBuffer : ASAPI_ServerMessage::Type::PlayingBuffer);
 }

Libraries/LibAudio/AClientConnection.h

@@ -11,5 +11,5 @@ public:
     AClientConnection();
 
     virtual void handshake() override;
-    void play(const ABuffer&);
+    void play(const ABuffer&, bool block);
 };

Libraries/LibAudio/ASAPI.h

@@ -5,6 +5,7 @@ struct ASAPI_ServerMessage {
         Invalid,
         Greeting,
         PlayingBuffer,
+        FinishedPlayingBuffer,
     };
 
     Type type { Type::Invalid };

Libraries/LibAudio/AWavLoader.cpp

@@ -2,26 +2,32 @@
 #include <LibAudio/ABuffer.h>
 #include <LibAudio/AWavLoader.h>
 #include <LibCore/CFile.h>
+#include <LibCore/CFileStreamReader.h>
 #include <limits>
 
-RefPtr<ABuffer> AWavLoader::load_wav(const StringView& path)
+AWavLoader::AWavLoader(const StringView& path)
+    : m_file(path)
 {
-    m_error_string = {};
+    if (!m_file.open(CIODevice::ReadOnly)) {
-
+        m_error_string = String::format("Can't open file: %s", m_file.error_string());
-    CFile wav(path);
+        return;
-    if (!wav.open(CIODevice::ReadOnly)) {
-        m_error_string = String::format("Can't open file: %s", wav.error_string());
-        return nullptr;
     }
 
-    auto contents = wav.read_all();
+    parse_header();
-    return parse_wav(contents);
 }
 
-// TODO: A streaming parser might be better than forcing a ByteBuffer
+RefPtr<ABuffer> AWavLoader::get_more_samples()
-RefPtr<ABuffer> AWavLoader::parse_wav(ByteBuffer& buffer)
 {
-    BufferStream stream(buffer);
+    dbgprintf("Read WAV of format PCM with num_channels %u sample rate %u, bits per sample %u\n", m_num_channels, m_sample_rate, m_bits_per_sample);
+
+    auto raw_samples = m_file.read(128 * KB);
+    auto buffer = ABuffer::from_pcm_data(raw_samples, m_num_channels, m_bits_per_sample, m_sample_rate);
+    return buffer;
+}
+
+bool AWavLoader::parse_header()
+{
+    CFileStreamReader stream(m_file);
 
 #define CHECK_OK(msg)                                                           \
     do {                                                                        \
@@ -73,13 +79,11 @@ RefPtr<ABuffer> AWavLoader::parse_wav(ByteBuffer& buffer)
     ASSERT(audio_format == 1);
     CHECK_OK("Audio format"); // value check
 
-    u16 num_channels;
+    stream >> m_num_channels;
-    stream >> num_channels;
+    ok = ok && (m_num_channels == 1 || m_num_channels == 2);
-    ok = ok && (num_channels == 1 || num_channels == 2);
     CHECK_OK("Channel count");
 
-    u32 sample_rate;
+    stream >> m_sample_rate;
-    stream >> sample_rate;
     CHECK_OK("Sample rate");
 
     u32 byte_rate;
@@ -90,11 +94,10 @@ RefPtr<ABuffer> AWavLoader::parse_wav(ByteBuffer& buffer)
     stream >> block_align;
     CHECK_OK("Block align");
 
-    u16 bits_per_sample;
+    stream >> m_bits_per_sample;
-    stream >> bits_per_sample;
     CHECK_OK("Bits per sample"); // incomplete read check
-    ok = ok && (bits_per_sample == 8 || bits_per_sample == 16 || bits_per_sample == 24);
+    ok = ok && (m_bits_per_sample == 8 || m_bits_per_sample == 16 || m_bits_per_sample == 24);
-    ASSERT(bits_per_sample == 8 || bits_per_sample == 16 || bits_per_sample == 24);
+    ASSERT(m_bits_per_sample == 8 || m_bits_per_sample == 16 || m_bits_per_sample == 24);
     CHECK_OK("Bits per sample"); // value check
 
     // Read chunks until we find DATA
@@ -116,21 +119,13 @@ RefPtr<ABuffer> AWavLoader::parse_wav(ByteBuffer& buffer)
     CHECK_OK("Found no data chunk");
     ASSERT(found_data);
 
-    ok = ok && data_sz < std::numeric_limits<int>::max();
+    ok = ok && data_sz < INT32_MAX;
     CHECK_OK("Data was too large");
 
-    // ### consider using BufferStream to do this for us
-    ok = ok && int(data_sz) <= (buffer.size() - stream.offset());
-    CHECK_OK("Bad DATA (truncated)");
-
     // Just make sure we're good before we read the data...
     ASSERT(!stream.handle_read_failure());
 
-    auto sample_data = buffer.slice_view(stream.offset(), data_sz);
+    return true;
-
-    dbgprintf("Read WAV of format PCM with num_channels %d sample rate %d, bits per sample %d\n", num_channels, sample_rate, bits_per_sample);
-
-    return ABuffer::from_pcm_data(sample_data, num_channels, bits_per_sample, sample_rate);
 }
 
 // Small helper to resample from one playback rate to another

Libraries/LibAudio/AWavLoader.h

@@ -3,6 +3,7 @@
 #include <AK/AKString.h>
 #include <AK/RefPtr.h>
 #include <AK/StringView.h>
+#include <LibCore/CFile.h>
 
 class ABuffer;
 
@@ -13,10 +14,18 @@ class ByteBuffer;
 // Parses a WAV file and produces an ABuffer instance from it
 class AWavLoader {
 public:
+    explicit AWavLoader(const StringView& path);
     RefPtr<ABuffer> load_wav(const StringView& path);
     const char* error_string() { return m_error_string.characters(); }
 
+    RefPtr<ABuffer> get_more_samples();
+
 private:
-    RefPtr<ABuffer> parse_wav(ByteBuffer&);
+    bool parse_header();
+    CFile m_file;
     String m_error_string;
+
+    u32 m_sample_rate { 0 };
+    u16 m_num_channels { 0 };
+    u16 m_bits_per_sample { 0 };
 };

Servers/AudioServer/ASClientConnection.cpp

@@ -48,11 +48,6 @@ bool ASClientConnection::handle_message(const ASAPI_ClientMessage& message, cons
                 did_misbehave();
                 return false;
             }
-
-            if (shared_buf->size() / sizeof(ASample) > 441000) {
-                did_misbehave();
-                return false;
-            }
             samples.resize(shared_buf->size() / sizeof(ASample));
             memcpy(samples.data(), shared_buf->data(), shared_buf->size());
         }
@@ -64,7 +59,7 @@ bool ASClientConnection::handle_message(const ASAPI_ClientMessage& message, cons
         reply.playing_buffer.buffer_id = message.play_buffer.buffer_id;
         post_message(reply);
 
-        m_mixer.queue(*this, adopt(*new ABuffer(move(samples))));
+        m_mixer.queue(*this, ABuffer::create_with_samples(move(samples)), message.play_buffer.buffer_id);
         break;
     }
     case ASAPI_ClientMessage::Type::Invalid:

Servers/AudioServer/ASMixer.cpp

@@ -1,8 +1,9 @@
 #include <AK/BufferStream.h>
+#include <AudioServer/ASClientConnection.h>
+#include <AudioServer/ASMixer.h>
+#include <LibAudio/ASAPI.h>
 #include <LibCore/CThread.h>
-
 #include <limits>
-#include "ASMixer.h"
 
 ASMixer::ASMixer()
     : m_device("/dev/audio")
@@ -19,11 +20,11 @@ ASMixer::ASMixer()
     }, this);
 }
 
-void ASMixer::queue(ASClientConnection&, const ABuffer& buffer)
+void ASMixer::queue(ASClientConnection& client, const ABuffer& buffer, int buffer_id)
 {
     ASSERT(buffer.size_in_bytes());
     CLocker lock(m_lock);
-    m_pending_mixing.append(ASMixerBuffer(buffer));
+    m_pending_mixing.append(ASMixerBuffer(buffer, client, buffer_id));
 }
 
 void ASMixer::mix()
@@ -75,8 +76,15 @@ void ASMixer::mix()
             }
 
             // clear it later
-            if (buffer.pos == samples.size())
+            if (buffer.pos == samples.size()) {
+                if (buffer.m_buffer_id && buffer.m_client) {
+                    ASAPI_ServerMessage reply;
+                    reply.type = ASAPI_ServerMessage::Type::FinishedPlayingBuffer;
+                    reply.playing_buffer.buffer_id = buffer.m_buffer_id;
+                    buffer.m_client->post_message(reply);
+                }
                 buffer.done = true;
+            }
         }
 
         // output the mixed stuff to the device
@@ -108,3 +116,10 @@ void ASMixer::mix()
         }
     }
 }
+
+ASMixer::ASMixerBuffer::ASMixerBuffer(const NonnullRefPtr<ABuffer>& buf, ASClientConnection& client, int buffer_id)
+    : buffer(buf)
+    , m_client(client.make_weak_ptr())
+    , m_buffer_id(buffer_id)
+{
+}

Servers/AudioServer/ASMixer.h

@@ -1,11 +1,12 @@
 #pragma once
 
-#include <AK/RefCounted.h>
 #include <AK/ByteBuffer.h>
+#include <AK/NonnullRefPtrVector.h>
+#include <AK/RefCounted.h>
+#include <AK/WeakPtr.h>
+#include <LibAudio/ABuffer.h>
 #include <LibCore/CFile.h>
 #include <LibCore/CLock.h>
-#include <LibAudio/ABuffer.h>
-#include <AK/NonnullRefPtrVector.h>
 
 class ASClientConnection;
 
@@ -13,16 +14,16 @@ class ASMixer : public RefCounted<ASMixer> {
 public:
     ASMixer();
 
-    void queue(ASClientConnection&, const ABuffer&);
+    void queue(ASClientConnection&, const ABuffer&, int buffer_id);
 
 private:
     struct ASMixerBuffer {
-        ASMixerBuffer(const NonnullRefPtr<ABuffer>& buf)
+        ASMixerBuffer(const NonnullRefPtr<ABuffer>&, ASClientConnection&, int buffer_id = -1);
-            : buffer(buf)
-        {}
         NonnullRefPtr<ABuffer> buffer;
         int pos { 0 };
         bool done { false };
+        WeakPtr<ASClientConnection> m_client;
+        int m_buffer_id { -1 };
     };
 
     Vector<ASMixerBuffer> m_pending_mixing;

Userland/aplay.cpp

@@ -16,15 +16,18 @@ int main(int argc, char **argv)
     AClientConnection a_conn;
     a_conn.handshake();
     printf("Established connection\n");
-    AWavLoader loader;
+    AWavLoader loader(argv[1]);
-    const auto& buffer = loader.load_wav(argv[1]);
+    printf("Loaded WAV\n");
-    if (!buffer) {
+
-        dbgprintf("Can't parse WAV: %s\n", loader.error_string());
+    for (;;) {
-        return 1;
+        auto samples = loader.get_more_samples();
+        if (!samples) {
+            break;
+        }
+        printf("Playing %d sample(s)\n", samples->samples().size());
+        a_conn.play(*samples, true);
     }
 
-    printf("Playing WAV\n");
-    a_conn.play(*buffer);
     printf("Exiting! :)\n");
     return 0;
 }