Dolby Surround decoding audio plugin

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3314 b3059339-0415-0410-9bf9-f77b7e298cf2

libao2/pl_surround.c

@@ -0,0 +1,177 @@
+/* 
+   This is an ao2 plugin to do simple decoding of matrixed surround
+   sound.  This will provide a (basic) surround-sound effect from
+   audio encoded for Dolby Surround, Pro Logic etc.
+
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+   Original author: Steve Davies <steve@daviesfam.org>
+*/
+
+/* The principle:  Make rear channels by extracting anti-phase data
+   from the front channels, delay by 15msec and feed to rear in anti-phase
+   www.dolby.com has the background
+*/
+
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "audio_out.h"
+#include "audio_plugin.h"
+#include "audio_plugin_internal.h"
+#include "afmt.h"
+
+static ao_info_t info =
+{
+        "Surround decoder plugin",
+        "surround",
+        "Steve Davies <steve@daviesfam.org>",
+        ""
+};
+
+LIBAO_PLUGIN_EXTERN(surround)
+
+// local data
+typedef struct pl_surround_s
+{
+  int passthrough;      // Just be a "NO-OP"
+  int msecs;            // Rear channel delay in milliseconds
+  int16_t* databuf;     // Output audio buffer
+  int16_t* delaybuf;    // circular buffer to be used for delaying audio signal
+  int delaybuf_len;     // local buffer length in samples
+  int delaybuf_ptr;     // offset in buffer where we are reading/writing
+  int rate;             // input data rate
+  int format;           // input format
+  int input_channels;   // input channels
+
+} pl_surround_t;
+
+static pl_surround_t pl_surround={0,15,NULL,NULL,0,0,0,0,0};
+
+// to set/get/query special features/parameters
+static int control(int cmd,int arg){
+  switch(cmd){
+  case AOCONTROL_PLUGIN_SET_LEN:
+    if (pl_surround.passthrough) return CONTROL_OK;
+    //fprintf(stderr, "pl_surround: AOCONTROL_PLUGIN_SET_LEN with arg=%d\n", arg);
+    //fprintf(stderr, "pl_surround: ao_plugin_data.len=%d\n", ao_plugin_data.len);
+    // Allocate an output buffer
+    if (pl_surround.databuf != NULL) {
+      free(pl_surround.databuf);  pl_surround.databuf = NULL;
+    }
+    pl_surround.databuf = calloc(ao_plugin_data.len, 1);
+    // Return back smaller len so we don't get overflowed...  (??seems the right thing to do?)
+    ao_plugin_data.len /= 2;
+    return CONTROL_OK;
+  }
+  return -1;
+}
+
+// open & setup audio device
+// return: 1=success 0=fail
+static int init(){
+
+  fprintf(stderr, "pl_surround: init input rate=%d, channels=%d\n", ao_plugin_data.rate, ao_plugin_data.channels);
+  if (ao_plugin_data.channels != 2) {
+    fprintf(stderr, "pl_surround: source audio must have 2 channels, using passthrough mode\n");
+    pl_surround.passthrough = 1;
+    return 1;
+  }
+  if (ao_plugin_data.format != AFMT_S16_LE) {
+    fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_LE audio format, using passthrough mode\n");
+    pl_surround.passthrough = 1;
+    return 1;
+  }
+
+  pl_surround.passthrough = 0;
+
+  /* Store info on input format to expect */
+  pl_surround.rate=ao_plugin_data.rate;
+  pl_surround.format=ao_plugin_data.format;
+  pl_surround.input_channels=ao_plugin_data.channels;
+
+  // Input 2 channels, output will be 4 - tell ao_plugin
+  ao_plugin_data.channels    = 4;
+  ao_plugin_data.sz_mult    /= 2;
+
+  // Figure out buffer space needed for the 15msec delay
+  pl_surround.delaybuf_len = pl_surround.rate * pl_surround.msecs / 1000;
+  // Allocate delay buffer
+  pl_surround.delaybuf=(void*)calloc(pl_surround.delaybuf_len,sizeof(int16_t));
+  fprintf(stderr, "pl_surround: %dmsec surround delay, rate %d - buffer is %d samples\n",
+	  pl_surround.msecs,pl_surround.rate,  pl_surround.delaybuf_len);
+  pl_surround.delaybuf_ptr = 0;
+
+  return 1;
+}
+
+// close plugin
+static void uninit(){
+  //  fprintf(stderr, "pl_surround: uninit called!\n");
+  if (pl_surround.passthrough) return;
+  if(pl_surround.delaybuf) 
+    free(pl_surround.delaybuf);
+  if(pl_surround.databuf) 
+    free(pl_surround.databuf);
+  pl_surround.delaybuf_len=0;
+}
+
+// empty buffers
+static void reset()
+{
+  if (pl_surround.passthrough) return;
+  //fprintf(stderr, "pl_surround: reset called\n");
+  pl_surround.delaybuf_ptr = 0;
+  memset(pl_surround.delaybuf, 0, sizeof(int16_t)*pl_surround.delaybuf_len);
+}
+
+
+// processes 'ao_plugin_data.len' bytes of 'data'
+// called for every block of data
+static int play(){
+  int16_t *in, *out;
+  int i, samples;
+  int surround;
+
+  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;
+  for (i=0; i<samples; i++) {
+    // front left and right
+    out[0] = in[0];
+    out[1] = in[1];
+    // surround - from 15msec ago
+    out[2] = pl_surround.delaybuf[pl_surround.delaybuf_ptr];
+    out[3] = -out[2];
+    // calculate and save surround for 15msecs time
+    pl_surround.delaybuf[pl_surround.delaybuf_ptr++] = (in[0]/2 - in[1]/2);
+    pl_surround.delaybuf_ptr %= pl_surround.delaybuf_len;
+    // next samples...
+    in = &in[pl_surround.input_channels];  out = &out[4];
+  }
+  
+  // Set output block/len
+  ao_plugin_data.data=pl_surround.databuf;
+  ao_plugin_data.len=samples*sizeof(int16_t)*4;
+  return 1;
+}
+
+
+