Bug fix for AAC streaming.

src/net/majorkernelpanic/streaming/audio/AACStream.java

@@ -92,12 +92,16 @@ public AACStream() throws IOException {
 		if (!AACStreamingSupported()) {
 			Log.e(TAG,"AAC not supported on this phone");
 			throw new AACNotSupportedException();
+		} else {
+			Log.d(TAG,"AAC supported on this phone");
 		}
 
-		if (mMode == MODE_MEDIARECORDER_API) 
+		if (mMode == MODE_MEDIARECORDER_API) {
+			mQuality = new AudioQuality(16000,32000);
 			mPacketizer = new AACADTSPacketizer();
-		else 
+		} else { 
 			mPacketizer = new AACLATMPacketizer();
+		}
 
 	}
 
@@ -125,16 +129,9 @@ public void start() throws IllegalStateException, IOException {
 	}
 
 	@Override
-	@SuppressLint("InlinedApi")
 	protected void encodeWithMediaRecorder() throws IOException {
 		testADTS();
 		((AACADTSPacketizer)mPacketizer).setSamplingRate(mActualSamplingRate);
-		setAudioEncoder(MediaRecorder.AudioEncoder.AAC);
-		try {
-			Field name = MediaRecorder.OutputFormat.class.getField("AAC_ADTS");
-			setOutputFormat(name.getInt(null));
-		}
-		catch (Exception ignore) {}
 		super.encodeWithMediaRecorder();
 	}
 
@@ -274,8 +271,31 @@ public String generateSessionDescription() throws IllegalStateException, IOExcep
 	 * @throws IOException 
 	 * @throws IllegalStateException
 	 */
+	@SuppressLint("InlinedApi")
 	private void testADTS() throws IllegalStateException, IOException {
 
+		setAudioEncoder(MediaRecorder.AudioEncoder.AAC);
+		try {
+			Field name = MediaRecorder.OutputFormat.class.getField("AAC_ADTS");
+			setOutputFormat(name.getInt(null));
+		}
+		catch (Exception ignore) {
+			setOutputFormat(6);
+		}
+
+		// Checks if the user has supplied an exotic sampling rate
+		int i=0;
+		for (;i<AUDIO_SAMPLING_RATES.length;i++) {
+			if (AUDIO_SAMPLING_RATES[i] == mQuality.samplingRate) {
+				break;
+			}
+		}
+		// If he did, we force a reasonable one: 24 kHz
+		if (i>12) {
+			Log.e(TAG,"Not a valid sampling rate: "+mQuality.samplingRate);
+			mQuality.samplingRate = 24000;
+		}
+
 		if (mSettings!=null) {
 			if (mSettings.contains("aac-"+mQuality.samplingRate)) {
 				String[] s = mSettings.getString("aac-"+mQuality.samplingRate, "").split(",");
@@ -305,13 +325,14 @@ private void testADTS() throws IllegalStateException, IOException {
 		mMediaRecorder.setAudioSamplingRate(mQuality.samplingRate);
 		mMediaRecorder.setAudioEncodingBitRate(mQuality.bitRate);
 		mMediaRecorder.setOutputFile(TESTFILE);
+		mMediaRecorder.setMaxDuration(1000);
 		mMediaRecorder.prepare();
 		mMediaRecorder.start();
 
 		// We record for 1 sec
 		// TODO: use the MediaRecorder.OnInfoListener
 		try {
-			Thread.sleep(1000);
+			Thread.sleep(2000);
 		} catch (InterruptedException e) {}
 
 		mMediaRecorder.stop();
@@ -331,7 +352,7 @@ private void testADTS() throws IllegalStateException, IOException {
 
 		raf.read(buffer,1,5);
 
-		mSamplingRateIndex = (buffer[1]&0x3C) >> 2;
+		mSamplingRateIndex = (buffer[1]&0x3C)>>2 ;
 		mProfile = ( (buffer[1]&0xC0) >> 6 ) + 1 ;
 		mChannel = (buffer[1]&0x01) << 2 | (buffer[2]&0xC0) >> 6 ;
 		mActualSamplingRate = AUDIO_SAMPLING_RATES[mSamplingRateIndex];

src/net/majorkernelpanic/streaming/rtp/H264Packetizer.java

@@ -45,7 +45,9 @@ public class H264Packetizer extends AbstractPacketizer implements Runnable {
 	private long delay = 0, oldtime = 0;
 	private Statistics stats = new Statistics();
 	private byte[] sps = null, pps = null;
-
+	private int count = 0;
+
+
 	public H264Packetizer() throws IOException {
 		super();
 		socket.setClockFrequency(90000);
@@ -77,6 +79,8 @@ public void run() {
 		long duration = 0, delta2 = 0;
 		Log.d(TAG,"H264 packetizer started !");
 		stats.reset();
+		count = 0;
+
 		// This will skip the MPEG4 header if this step fails we can't stream anything :(
 		try {
 			byte buffer[] = new byte[4];
@@ -163,8 +167,13 @@ private void send() throws IOException, InterruptedException {
 
 		// The stream already contains NAL unit type 7 or 8, we don't need 
 		// to add them to the stream ourselves
-		if (type == 7) sps = null;
-		if (type == 8) pps = null;
+		if (type == 7 || type == 8) {
+			count++;
+			if (count>4) {
+				sps = null;
+				pps = null;
+			}
+		}
 
 		// Updates the timestamp
 		ts += delay;

src/net/majorkernelpanic/streaming/rtp/RtpSocket.java

@@ -227,7 +227,7 @@ public void run() {
 					if ((mTimestamps[mBufferOut]-mOldTimestamp)>0) {
 						stats.push(mTimestamps[mBufferOut]-mOldTimestamp);
 						long d = stats.average()/1000000;
-						Log.d(TAG,"delay: "+d+" d: "+(mTimestamps[mBufferOut]-mOldTimestamp)/1000000);
+						//Log.d(TAG,"delay: "+d+" d: "+(mTimestamps[mBufferOut]-mOldTimestamp)/1000000);
 						// We ensure that packets are sent at a constant and suitable rate no matter how the RtpSocket is used.
 						Thread.sleep(d);
 					}

src/net/majorkernelpanic/streaming/video/VideoStream.java

@@ -76,7 +76,6 @@ public VideoStream(int camera) {
 		setCamera(camera);
 		// TODO: Remove this when encoding with the MediaCodec API is ready
 		setMode(MODE_MEDIARECORDER_API);
-		Log.e(TAG,"MODE: "+mMode);		
 	}
 
 	/**