Merge

--HG--
branch : nfb

KREngine/kraken/KRAudioManager.cpp

@@ -406,6 +406,9 @@ void KRAudioManager::renderBlock()
         
         // ----====---- Render Ambient Sound ----====----
         renderAmbient();
+        
+        // ----====---- Render Ambient Sound ----====----
+        renderLimiter();
     }
     
     // ----====---- Advance audio sources ----====----
@@ -1488,7 +1491,19 @@ void KRAudioManager::setGlobalAmbientGain(float gain)
 
 void KRAudioManager::startFrame(float deltaTime)
 {
-    m_mutex.lock();
+    static unsigned long trackCount = 0;
+    static unsigned long trackMissed = 0;
+    trackCount++;
+    if (trackCount > 200) {
+//        printf("Missed %ld out of 200 try_lock attempts on audio startFrame\n", trackMissed);
+        trackCount = 0;
+        trackMissed = 0;
+    }
+    
+    if (!m_mutex.try_lock()) {
+        trackMissed++;
+        return;     // if we are rendering audio don't update audio state
+    }               // NOTE: this misses anywhere from 0 to to 30 times out of 200 on the iPad2
     
     // ----====---- Determine Ambient Zone Contributions ----====----
     m_ambient_zone_weights.clear();
@@ -1932,3 +1947,113 @@ void KRAudioManager::renderITD()
      
      */
 }
+
+static bool audioIsMuted = false;
+static bool audioShouldBecomeMuted = false;
+static bool audioShouldBecomeUnmuted = false;
+
+void audioLimit_Mute(bool onNotOff) {
+    if (onNotOff) {
+        if (audioIsMuted) return;
+        audioShouldBecomeMuted = true;
+        audioShouldBecomeUnmuted = false;
+    }
+    else {
+        if (!audioIsMuted) return;
+        audioShouldBecomeMuted = false;
+        audioShouldBecomeUnmuted = true;
+    }
+}
+
+float audioGetLimitParameters_Stereo(float *buffer, unsigned long framesize,
+                                  unsigned long *attack_sample_position, float *peak)
+{
+	float limitvol = 1.0;
+	long attack_position = -1;
+	*peak = 0.0;
+	float max = 0.0;
+	float amplitude = 0.0;
+	
+	float *src = buffer;
+	for (unsigned long i = 0; i < framesize * 2; i++) {
+		amplitude = fabs(*src); src++;
+		if (amplitude > max) max = amplitude;
+		if (amplitude > 0.995) if (attack_position < 0) attack_position = (i+1) / 2;
+    }
+	if (max > 0.995) limitvol = 0.995 / max;
+	*peak = max;
+	
+	if (attack_position < 0) attack_position = framesize;
+	*attack_sample_position = (unsigned long) attack_position;
+	return limitvol;
+} // returns the new limit volume, *attack_sample_position tells how fast we need to reach the new limit
+
+void audioLimit_Stereo(float *stereo_buffer, unsigned long framesize)
+{
+    static float limit_value = 1.0;
+
+    // (1) get the limiting parameters for the incoming audio data
+	float previouslimitvol = limit_value;
+	float peak;
+	unsigned long attack_sample_position = framesize;
+
+    // (1a) check for a mute or unmute state then get the next limit volume
+    float nextlimitvol = 0.0;
+    if (audioIsMuted && audioShouldBecomeUnmuted) { audioIsMuted = false; audioShouldBecomeUnmuted = false; }
+    if (audioShouldBecomeMuted) { audioIsMuted = true; audioShouldBecomeMuted = false; }
+	if (!audioIsMuted) nextlimitvol = audioGetLimitParameters_Stereo(stereo_buffer, framesize, &attack_sample_position, &peak);
+
+    // (1b) if no limiting is needed then return
+	if ((1.0 == nextlimitvol) && (1.0 == previouslimitvol)) { return; }	// no limiting necessary
+    
+    // (2) calculate limiting factors
+	float deltavol = 0.0;
+	if (previouslimitvol != nextlimitvol) {
+		deltavol = (nextlimitvol - previouslimitvol) / (float) attack_sample_position;
+    }
+
+    // (3) do the limiting
+	float *src = stereo_buffer;
+    
+	if (0.0 == deltavol) {	// fixed volume
+		for (unsigned long i=0; i < framesize; i++) {
+			*src = *src * nextlimitvol;
+            src++;
+			*src = *src * nextlimitvol;
+            src++;
+        }
+    }
+	else {
+		for (unsigned long i=0; i < attack_sample_position; i++) {	// attack phase
+			*src = *src * previouslimitvol;
+			src++;
+			*src = *src * previouslimitvol;
+			src++;
+			previouslimitvol += deltavol;
+        }
+		if (nextlimitvol < 1.0) {	// plateau phase
+			for (unsigned long i = attack_sample_position; i < framesize; i++) {
+				*src = *src * nextlimitvol;
+				src++;
+				*src = *src * nextlimitvol;
+				src++;
+            }
+        }
+    }
+        
+    // (4) save our limit level for next time
+    limit_value = nextlimitvol;
+}
+
+void KRAudioManager::mute(bool onNotOff)
+{
+    audioLimit_Mute(onNotOff);
+}
+
+void KRAudioManager::renderLimiter()
+{
+    int output_offset = (m_output_accumulation_block_start) % (KRENGINE_REVERB_MAX_SAMPLES * KRENGINE_MAX_OUTPUT_CHANNELS);
+    float *output = m_output_accumulation + output_offset;
+    unsigned long numframes = KRENGINE_AUDIO_BLOCK_LENGTH;
+    audioLimit_Stereo(output, numframes);
+}

KREngine/kraken/KRAudioManager.h

@@ -40,13 +40,28 @@
 #include "KRMat4.h"
 #include "KRAudioSource.h"
 
-const int KRENGINE_AUDIO_MAX_POOL_SIZE = 32;
+const int KRENGINE_AUDIO_MAX_POOL_SIZE = 60; //32;
-const int KRENGINE_AUDIO_MAX_BUFFER_SIZE = 64*1024;
+    // for Circa we play a maximum of 11 mono audio streams at once + cross fading with ambient
+    // so we could safely say a maximum of 12 or 13 streams, which would be 39 buffers
+    // do the WAV files for the reverb use the same buffer pool ???
+
+const int KRENGINE_AUDIO_MAX_BUFFER_SIZE = 5120;  // in bytes
+    // this is the buffer for our decoded audio (not the source file data)
+    // it should be greater then 1152 samples (the size of an mp3 frame in samples)
+    // so it should be greater then 2304 bytes and also a multiple of 128 samples (to make
+    // the data flow efficient) but it shouldn't be too large or it will cause
+    // the render loop to stall out decoding large chunks of mp3 data.
+    // 2560 bytes would be the smallest size for mono sources, and 5120 would be smallest for stereo.
+
 const int KRENGINE_AUDIO_BUFFERS_PER_SOURCE = 3;
 
-const int KRENGINE_AUDIO_BLOCK_LENGTH = 128; // Length of one block to process.  Determines the latency of the audio system and sets size for FFT's used in HRTF convolution
-
 const int KRENGINE_AUDIO_BLOCK_LOG2N = 7;   // 2 ^ KRENGINE_AUDIO_BLOCK_LOG2N = KRENGINE_AUDIO_BLOCK_LENGTH
+    // 7 is 128 .. NOTE: the hrtf code uses magic numbers everywhere and is hardcoded to 128 samples per frame
+
+const int KRENGINE_AUDIO_BLOCK_LENGTH = 1 << KRENGINE_AUDIO_BLOCK_LOG2N;
+    // Length of one block to process.  Determines the latency of the audio system and sets size for FFT's used in HRTF convolution
+    // the AUGraph works in 1024 sample chunks. At 128 we are making 8 consecutive calls to the renderBlock method for each
+    // render initiated by the AUGraph.
 
 const int KRENGINE_REVERB_MAX_FFT_LOG2 = 15;
 const int KRENGINE_REVERB_WORKSPACE_SIZE = 1 << KRENGINE_REVERB_MAX_FFT_LOG2;
@@ -128,6 +143,7 @@ public:
     
     KRAudioBuffer *getBuffer(KRAudioSample &audio_sample, int buffer_index);
     
+    static void mute(bool onNotOff);
 
     void startFrame(float deltaTime);
     
@@ -216,6 +232,7 @@ private:
     void renderHRTF();
     void renderITD();
     void renderReverbImpulseResponse(int impulse_response_offset, int frame_count_log2);
+    void renderLimiter();
     
     std::vector<KRVector2> m_hrtf_sample_locations;
     float *m_hrtf_data;

KREngine/kraken/KRTextureManager.cpp

@@ -329,6 +329,10 @@ void KRTextureManager::balanceTextureMemory()
         }
     }
     
+    if (maxDimActive > 512) maxDimActive = 512;     // FINDME - hack to stop the texture resizer from blowing out its brains ..
+                                                    // when the texture quality goes up to 1024, the app runs for about 2 mins and then
+                                                    // runs out of memory.
+    
     // Resize active textures to balance the memory usage and mipmap levels
     for(std::set<KRTexture *>::iterator itr=m_activeTextures_streamer.begin(); itr != m_activeTextures_streamer.end() && memory_available > 0; itr++) {
         KRTexture *activeTexture = *itr;