fix crackling/popping when playing music on openal backend

SharpWave/Backends/IAudioOutput.cs

@@ -27,12 +27,33 @@ namespace SharpWave {
 		
 		public AudioFormat Format;
 		public abstract void SetFormat(AudioFormat format);
-		public abstract void PlayData(int index, AudioChunk chunk);
+		public abstract void BufferData(int index, AudioChunk chunk);
+		public abstract void Play();
 		public abstract bool IsCompleted(int index);
 		public abstract bool IsFinished();
 		
 		public int NumBuffers;
 		public bool pendingStop;
+		public void PlayData(int index, AudioChunk chunk) {
+			BufferData(index, chunk);
+			Play();
+		}
+		
+		bool BufferBlock(AudioChunk tmp, AudioFormat fmt, IEnumerator<AudioChunk> chunks) {
+			// decode up to around a second
+			int secondSize = fmt.SampleRate * fmt.Channels * sizeof(short);
+			tmp.Length = 0;			
+			
+			while (tmp.Length < secondSize) {
+				if (!chunks.MoveNext()) return true;
+				AudioChunk src = chunks.Current;
+				
+				Buffer.BlockCopy(src.Data, 0, tmp.Data, tmp.Length, src.Length);
+				tmp.Length += src.Length;
+			}
+			return false;
+		}
+		
 		public void PlayStreaming(Stream stream) {
 			VorbisCodec codec = new VorbisCodec();
 			AudioFormat fmt = codec.ReadHeader(stream);
@@ -43,12 +64,19 @@ namespace SharpWave {
 			
 			// largest possible vorbis frame decodes to blocksize1 samples
 			// so we may end up decoding slightly over a second of audio
-			int secondSize = fmt.SampleRate         * fmt.Channels * sizeof(short);
 			int chunkSize = (fmt.SampleRate + 8192) * fmt.Channels * sizeof(short);
 			byte[][] data = new byte[NumBuffers][];
-			for (int i = 0; i < NumBuffers; i++) { data[i] = new byte[chunkSize]; }			
+			for (int i = 0; i < NumBuffers; i++) { data[i] = new byte[chunkSize]; }
 			
-			for (;;) {
+			bool reachedEnd = false;
+			for (int i = 0; i < NumBuffers && !reachedEnd; i++) {
+				tmp.Data = data[i];
+				reachedEnd = BufferBlock(tmp, fmt, chunks);
+				BufferData(i, tmp);
+			}
+			Play();
+			
+			for (; !reachedEnd;) {
 				int next = -1;
 				for (int i = 0; i < NumBuffers; i++) {
 					if (IsCompleted(i)) { next = i; break; }
@@ -56,23 +84,10 @@ namespace SharpWave {
 
 				if (next == -1) { Thread.Sleep(10); continue; }
 				if (pendingStop) break;
-				
-				// decode up to around a second
+
 				tmp.Data = data[next];
-				tmp.Length = 0;
-				bool reachedEnd = false;
-
-				while (tmp.Length < secondSize) {
-					if (!chunks.MoveNext()) { reachedEnd = true; break; }
-
-					AudioChunk src = chunks.Current;
-					Buffer.BlockCopy(src.Data, 0, tmp.Data, tmp.Length, src.Length);
-					tmp.Length += src.Length;
-				}
-				
-				PlayData(next, tmp);
-				// need to specially handle last bit of audio
-				if (reachedEnd) break;
+				reachedEnd = BufferBlock(tmp, fmt, chunks);
+				BufferData(next, tmp);
 			}
 
 			while (!IsFinished()) { Thread.Sleep(10); }

SharpWave/Backends/OpenALOut.cs

@@ -48,12 +48,12 @@ namespace SharpWave {
 		
 		public override bool IsCompleted(int index) {
 			int processed = 0;
-			uint buffer = 0;			
+			uint buffer = 0;
 			lock (contextLock) {
 				MakeContextCurrent();
 				
 				AL.alGetSourcei(source, ALGetSourcei.BuffersProcessed, &processed);
-				CheckError("GetSources");			
+				CheckError("GetSources");
 				if (processed == 0) return completed[index];
 				
 				AL.alSourceUnqueueBuffers(source, 1, &buffer);
@@ -80,7 +80,7 @@ namespace SharpWave {
 			}
 		}
 		
-		public override void PlayData(int index, AudioChunk chunk) {
+		public override void BufferData(int index, AudioChunk chunk) {
 			fixed (byte* data = chunk.Data) {
 				uint buffer = bufferIDs[index];
 				completed[index] = false;
@@ -88,17 +88,23 @@ namespace SharpWave {
 				lock (contextLock) {
 					MakeContextCurrent();
 					AL.alBufferData(buffer, dataFormat, (IntPtr)data,
-					              chunk.Length, Format.SampleRate);
+					                chunk.Length, Format.SampleRate);
 					CheckError("BufferData");
 					
 					AL.alSourceQueueBuffers(source, 1, &buffer);
-					CheckError("QueueBuffers");				
-					AL.alSourcePlay(source);
-					CheckError("SourcePlay");
+					CheckError("QueueBuffers");
 				}
 			}
 		}
 		
+		public override void Play() {
+			lock (contextLock) {
+				MakeContextCurrent();
+				AL.alSourcePlay(source);
+				CheckError("SourcePlay");
+			}
+		}
+		
 		void CheckError(string location) {
 			ALError error = AL.alGetError();
 			if (error != ALError.NoError) {

SharpWave/Backends/WinMmOut.cs

@@ -70,7 +70,7 @@ namespace SharpWave {
 			CheckError(result, "Open");
 		}
 		
-		public override void PlayData(int index, AudioChunk chunk) {
+		public override void BufferData(int index, AudioChunk chunk) {
 			if (chunk.Length > dataSizes[index]) {
 				IntPtr ptr = dataHandles[index];
 				if (ptr != IntPtr.Zero) Marshal.FreeHGlobal(ptr);
@@ -97,6 +97,8 @@ namespace SharpWave {
 			CheckError(result, "Write");
 		}
 		
+		public override void Play() { }
+		
 		void ApplyVolume(IntPtr handle, AudioChunk chunk) {
 			if (volumePercent == 100) return;
 			

SharpWave/csvorbis/VorbisCodec.cs

@@ -11,11 +11,9 @@ namespace SharpWave {
 			chunk = new AudioChunk();
 		}
 
-		AudioChunk chunk, rawChunk;
+		AudioChunk chunk;
 		Stream input;
-		byte[] rawPcm;
-		int rawIndex;
-		
+
 		SyncState oy = new SyncState(); // sync and verify incoming physical bitstream
 		StreamState os = new StreamState(); // take physical pages, weld into a logical stream of packets
 		Page og = new Page(); // one Ogg bitstream page.  Vorbis packets are inside

src/Client/PacketHandlers.c

@@ -932,6 +932,7 @@ static void CPE_MakeSelection(UInt8* data) {
 	p2.Z = (Int16)Stream_GetU16_BE(&data[4]);
 
 	PackedCol c; data += 6;
+	/* R,G,B,A are actually 16 bit unsigned integers */
 	c.R = data[1]; c.G = data[3]; c.B = data[5]; c.A = data[7];
 	Selections_Add(selectionId, p1, p2, c);
 }