SoundPlayer: Handle any input file sample rate

This commit addresses two issues:
1. If you play a 96 KHz Wave file, the slider position is incorrect,
   because it is assumed all files are 44.1 KHz.
2. For high-bitrate files, there are audio dropouts due to not
   buffering enough audio data.

Issue 1 is addressed by scaling the number of played samples by the
ratio between the source and destination sample rates.

Issue 2 is addressed by buffering a certain number of milliseconds
worth of audio data (instead of a fixed number of bytes).
This makes the the buffer size independent of the source sample rate.

Some of the code is redesigned to be simpler. The code that did the
book-keeping of which buffers need to be loaded and which have been
already played has been removed. Instead, we enqueue a new buffer based
on a low watermark of samples remaining in the audio server queue.

Other small fixes include:
1. Disable the stop button when playback is finished.
2. Remove hard-coded instances of 44100.
3. Update the GUI every 50 ms (was 100), which improves visualizations.

Userland/Libraries/LibAudio/ClientConnection.cpp

@@ -20,7 +20,7 @@ void ClientConnection::enqueue(const Buffer& buffer)
         auto success = enqueue_buffer(buffer.anonymous_buffer(), buffer.id(), buffer.sample_count());
         if (success)
             break;
-        sleep(.1);
+        usleep(100000);
     }
 }
 

Userland/Libraries/LibAudio/WavLoader.cpp

@@ -30,7 +30,7 @@ WavLoaderPlugin::WavLoaderPlugin(const StringView& path)
     if (!valid)
         return;
 
-    m_resampler = make<ResampleHelper>(m_sample_rate, 44100);
+    m_resampler = make<ResampleHelper>(m_sample_rate, m_device_sample_rate);
 }
 
 WavLoaderPlugin::WavLoaderPlugin(const ByteBuffer& buffer)
@@ -46,7 +46,7 @@ WavLoaderPlugin::WavLoaderPlugin(const ByteBuffer& buffer)
     if (!valid)
         return;
 
-    m_resampler = make<ResampleHelper>(m_sample_rate, 44100);
+    m_resampler = make<ResampleHelper>(m_sample_rate, m_device_sample_rate);
 }
 
 RefPtr<Buffer> WavLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_input)

Userland/Libraries/LibAudio/WavLoader.h

@@ -41,6 +41,8 @@ public:
     virtual bool has_error() override { return !m_error_string.is_null(); }
     virtual const char* error_string() override { return m_error_string.characters(); }
 
+    // The Buffer returned contains input data resampled at the
+    // destination audio device sample rate.
     virtual RefPtr<Buffer> get_more_samples(size_t max_bytes_to_read_from_input = 128 * KiB) override;
 
     virtual void reset() override { return seek(0); }
@@ -64,6 +66,11 @@ private:
     OwnPtr<AK::InputStream> m_stream;
     AK::InputMemoryStream* m_memory_stream;
     String m_error_string;
+
+    // TODO: We should probably move resampling into the audio server.
+    //
+    // It would avoid duplicate resampling code and would allow clients
+    // to be agnostic of the destination audio device's sample rate.
     OwnPtr<ResampleHelper> m_resampler;
 
     u32 m_sample_rate { 0 };
@@ -71,6 +78,8 @@ private:
     PcmSampleFormat m_sample_format;
     size_t m_byte_offset_of_data_samples { 0 };
 
+    // FIXME: Get this value from the audio server
+    int m_device_sample_rate { 44100 };
     int m_loaded_samples { 0 };
     int m_total_samples { 0 };
 };