[MPlayer-dev-eng] [PATCH] current pl_surround in mplayer CVS breaks playlist

Kis Gergely kisg at lme.linux.hu
Sat Apr 6 22:55:19 CEST 2002


2002-04-06, szo keltezéssel Felix Buenemann ezt írta:
> On Saturday 06 April 2002 15:41, Kis Gergely wrote:
> > Actually it tries to free() an already freed buffer in
> > pl_surround.control(). The attached patch solves the problem.
> thx, commited.
> 
> Btw. is the extrasurround stuff now stable and ready to be commited? What is 
> the last patch for it?
Well, I currently fight with ao_plugin.c

I attach three patches.

The first patch fixes a bug in ao_plugin.c. ao_plugin_local_data.len was
not set to 0 on init and on uninit. This caused problems when using
playlists.

The second patch is the newest extrasurround patch (pl_surround.c). 
Changes:
->updated to current pl_surround.c
->subwoofer cutoff freq is set to 300 Hz. (After consulting with a more
competent friend of mine.)

If you compile mplayer, and try it out, it won't work. I tried to
compile both in debug and normal mode, with gcc 2.95.4 (Debian
prerelease) and gcc 3.0.4

The problem is the previously described "real databuf len" - 1 bug.

There is a workaround for that (look for Anders' mail).

BUT:
If you now apply the third patch, which only adds some fprintfs for
debug purposes and recompile, it works!!!

Now that's an X-file for me. I must be lame, or just found a compiler
bug in both 2.95.4 and 3.0.4... I think the first version is more
likely...:-(

Could somebody please enlighten me?
Or at least reproduce?


Thanks,
kisg


-------------- next part --------------
--- ao_plugin.c.orig	Fri Apr  5 10:16:13 2002
+++ ao_plugin.c	Sat Apr  6 21:21:31 2002
@@ -172,6 +172,7 @@
   if(ao_plugin_local_data.buf)
     free(ao_plugin_local_data.buf);
   ao_plugin_local_data.buf=malloc(MAX_OUTBURST);
+  ao_plugin_local_data.len=0;
 
   if(!ao_plugin_local_data.buf) return 0;
 
@@ -190,6 +191,7 @@
   if(ao_plugin_local_data.buf)
     free(ao_plugin_local_data.buf);
   ao_plugin_local_data.buf=NULL;
+  ao_plugin_local_data.len=0;
 }
 
 // stop playing and empty buffers (for seeking/pause)
-------------- next part --------------
--- pl_surround.c.orig	Sat Apr  6 19:10:52 2002
+++ pl_surround.c	Sat Apr  6 22:53:17 2002
@@ -64,6 +64,7 @@
   int passthrough;      // Just be a "NO-OP"
   int msecs;            // Rear channel delay in milliseconds
   int16_t* databuf;     // Output audio buffer
+  int databuf_len;      // Output audio buffer length in samples
   int16_t* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
   int16_t* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
   int delaybuf_len;     // delaybuf buffer length in samples
@@ -72,10 +73,16 @@
   int rate;             // input data rate
   int format;           // input format
   int input_channels;   // input channels
+  int output_channels;  // output channels
+  float lowp_cutoff;    // cutoff freq for the lowpass filter
+  double lowp_A, lowp_B; // parameters for lowpass filter
+  double lowp_outm1;	// output for lowpass filter (??)
 
 } pl_surround_t;
 
-static pl_surround_t pl_surround={0,20,NULL,NULL,NULL,0,0,NULL,0,0,0};
+static pl_surround_t pl_surround={0,20,NULL,0,NULL,NULL,0,0,NULL,0,0,0,0,0,0,0,0};
+
+extern int audio_output_channels;
 
 // to set/get/query special features/parameters
 static int control(int cmd,int arg){
@@ -89,9 +96,20 @@
       free(pl_surround.databuf);  pl_surround.databuf = NULL;
     }
     // Allocate output buffer
-    pl_surround.databuf = calloc(ao_plugin_data.len, 1);
+    
+    fprintf(stderr, "pl_surround: We can produce: audio_plugin_data.len = %d\n",ao_plugin_data.len);
+    
+    pl_surround.databuf_len = ao_plugin_data.len / sizeof(int16_t) / pl_surround.output_channels;
+    
+    fprintf(stderr, "pl_surround: This means pl_surround.databuf_len = %d\n",pl_surround.databuf_len);
+    
+    pl_surround.databuf = calloc(pl_surround.databuf_len*pl_surround.output_channels, sizeof(int16_t));
     // Return back smaller len so we don't get overflowed...
-    ao_plugin_data.len /= 2;
+    // ao_plugin_data.len /= 2;
+    ao_plugin_data.len = pl_surround.databuf_len * sizeof(int16_t) * pl_surround.input_channels;
+
+    fprintf(stderr, "pl_surround: We can receive: audio_plugin_data.len = %d\n",ao_plugin_data.len);
+    
     return CONTROL_OK;
   }
   return -1;
@@ -113,6 +131,12 @@
     return 1;
   }
 
+  if (audio_output_channels != 4 && audio_output_channels != 6) {
+    fprintf(stderr, "pl_surround: I'm dumb and can only output 4 or 6 channel sound, using passtrough mode\n");
+    pl_surround.passthrough = 1;
+    return 1;    
+  }
+
   pl_surround.passthrough = 0;
 
   /* Store info on input format to expect */
@@ -120,9 +144,16 @@
   pl_surround.format=ao_plugin_data.format;
   pl_surround.input_channels=ao_plugin_data.channels;
 
+  /* Store info on output channel number */
+  pl_surround.output_channels = audio_output_channels;
+
+
   // Input 2 channels, output will be 4 - tell ao_plugin
-  ao_plugin_data.channels    = 4;
-  ao_plugin_data.sz_mult    /= 2;
+//  ao_plugin_data.channels    = 4;
+//  ao_plugin_data.sz_mult    /= 2;
+
+  ao_plugin_data.channels    = pl_surround.output_channels;
+  ao_plugin_data.sz_mult    /= pl_surround.output_channels / pl_surround.input_channels;
 
   // Figure out buffer space (in int16_ts) needed for the 15msec delay
   // Extra 31 samples allow for lowpass filter delay (taps-1)
@@ -137,6 +168,16 @@
   pl_surround.filter_coefs_surround = calc_coefficients_7kHz_lowpass(pl_surround.rate);
   //dump_filter_coefficients(pl_surround.filter_coefs_surround);
   //testfilter(pl_surround.filter_coefs_surround, 32, pl_surround.rate);
+  
+  pl_surround.lowp_cutoff = 300;
+  if (pl_surround.lowp_cutoff > pl_surround.rate / 2) {
+    // Cutoff rate must be < sample rate / 2 (Nyquist rate)
+    pl_surround.lowp_cutoff = pl_surround.rate / 2 - 1;
+  } 
+  pl_surround.lowp_B = exp ((-2.0 * M_PI * (pl_surround.lowp_cutoff / pl_surround.rate)));
+  pl_surround.lowp_A = 1 - pl_surround.lowp_B;
+  pl_surround.lowp_outm1 = 0.0;
+  
   return 1;
 }
 
@@ -163,6 +204,7 @@
   pl_surround.delaybuf_pos = 0;
   memset(pl_surround.Ls_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
   memset(pl_surround.Rs_delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
+  pl_surround.lowp_outm1 = 0;
 }
 
 // The beginnings of an active matrix...
@@ -183,11 +225,11 @@
 
   if (pl_surround.passthrough) return 1;
 
-  // fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
 
   samples  = ao_plugin_data.len / sizeof(int16_t) / pl_surround.input_channels;
   out = pl_surround.databuf;  in = (int16_t *)ao_plugin_data.data;
 
+  fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
   // Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
   //sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
   //sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
@@ -224,6 +266,26 @@
 #else
     out[3] = -out[2];
 #endif
+    if (pl_surround.output_channels == 6) {
+    
+	int32_t avg;
+	double d;
+	avg = (out[0] + out[1]) / 2;
+	
+	d = pl_surround.lowp_A * avg + pl_surround.lowp_B * pl_surround.lowp_outm1;
+
+	if (d > 32767L) {
+	    d = 32767L;
+	}
+	if (d < -32768L) {
+	    d = -32768L;
+	}
+	pl_surround.lowp_outm1 = d;
+	out[5] = d;
+	// Only 4.1 output, center speaker remains silent
+	out[4] = 0;
+    }
+    
     // calculate and save surround for 20msecs time
 #ifdef SPLITREAR
     pl_surround.Ls_delaybuf[pl_surround.delaybuf_pos] =
@@ -237,7 +299,7 @@
     pl_surround.delaybuf_pos %= pl_surround.delaybuf_len;
 
     // next samples...
-    in = &in[pl_surround.input_channels];  out = &out[4];
+    in = &in[pl_surround.input_channels];  out = &out[pl_surround.output_channels];
   }
 
   // Show some state
@@ -245,6 +307,6 @@
   
   // Set output block/len
   ao_plugin_data.data=pl_surround.databuf;
-  ao_plugin_data.len=samples*sizeof(int16_t)*4;
+  ao_plugin_data.len=samples*sizeof(int16_t)*pl_surround.output_channels;
   return 1;
 }
-------------- next part --------------
--- ao_plugin.c.bugfix1	Sat Apr  6 21:21:31 2002
+++ ao_plugin.c	Sat Apr  6 22:24:17 2002
@@ -216,10 +216,17 @@
 // return: how many bytes can be played without blocking
 static int get_space(){
   double sz=(double)(driver()->get_space());
-  if(sz+(double)ao_plugin_local_data.len > (double)MAX_OUTBURST)
+  fprintf(stderr,"ao_plugin: get_space(): Output driver says %f free space\n",sz);
+  fprintf(stderr,"ao_plugin: get_space(): ao_plugin_local_data.len says %f\n",(double)ao_plugin_local_data.len);
+  if(sz+(double)ao_plugin_local_data.len > (double)MAX_OUTBURST) {
+    fprintf(stderr,"ao_plugin: get_space(): sz+localbuffer length is > MAX_OUTBURST\n");
     sz=(double)MAX_OUTBURST-(double)ao_plugin_local_data.len;
+    fprintf(stderr,"ao_plugin: get_space(): sz=MAX_OUTBURST - localdata.len = %f\n",sz);
+  }
   sz*=ao_plugin_data.sz_mult;
+  fprintf(stderr,"ao_plugin: get_space(): sz*sz_mult =  %f\n",sz);
   sz+=ao_plugin_data.sz_fix;
+  fprintf(stderr,"ao_plugin: get_space(): sz+sz_fix =  %f\n",sz);
   return (int)(sz);
 }
 
@@ -230,6 +237,7 @@
   // Limit length to avoid over flow in plugins
   int tmp = get_space();
   int ret_len =(tmp<len)?tmp:len;
+  fprintf(stderr,"ao_plugin: play(): len = %d\n",len);
   if(ret_len){
     // Filter data
     ao_plugin_data.len=ret_len;


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