diff --git a/libmpcodecs/ad_pcm.c b/libmpcodecs/ad_pcm.c
index a5a17e1a49..b4d8a3f348 100644
--- a/libmpcodecs/ad_pcm.c
+++ b/libmpcodecs/ad_pcm.c
@@ -26,13 +26,12 @@
 #include "libaf/af_format.h"
 #include "libaf/reorder_ch.h"
 
-static const ad_info_t info =
-{
-	"Uncompressed PCM audio decoder",
-	"pcm",
-	"Nick Kurshev",
-	"A'rpi",
-	""
+static const ad_info_t info = {
+    "Uncompressed PCM audio decoder",
+    "pcm",
+    "Nick Kurshev",
+    "A'rpi",
+    ""
 };
 
 struct ad_pcm_context {
@@ -42,53 +41,55 @@ struct ad_pcm_context {
 
 LIBAD_EXTERN(pcm)
 
-static int init(sh_audio_t *sh_audio)
+static int init(sh_audio_t * sh_audio)
 {
-  WAVEFORMATEX *h=sh_audio->wf;
-  if (!h)
-    return 0;
-  sh_audio->i_bps=h->nAvgBytesPerSec;
-  sh_audio->channels=h->nChannels;
-  sh_audio->samplerate=h->nSamplesPerSec;
-  sh_audio->samplesize=(h->wBitsPerSample+7)/8;
-  sh_audio->sample_format=AF_FORMAT_S16_LE; // default
-  switch(sh_audio->format){ /* hardware formats: */
+    WAVEFORMATEX *h = sh_audio->wf;
+    if (!h)
+        return 0;
+    sh_audio->i_bps = h->nAvgBytesPerSec;
+    sh_audio->channels = h->nChannels;
+    sh_audio->samplerate = h->nSamplesPerSec;
+    sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
+    sh_audio->sample_format = AF_FORMAT_S16_LE; // default
+    switch (sh_audio->format) { /* hardware formats: */
     case 0x0:
-    case 0x1: // Microsoft PCM
-    case 0xfffe: // Extended
-       switch (sh_audio->samplesize) {
-         case 1: sh_audio->sample_format=AF_FORMAT_U8; break;
-         case 2: sh_audio->sample_format=AF_FORMAT_S16_LE; break;
-         case 3: sh_audio->sample_format=AF_FORMAT_S24_LE; break;
-         case 4: sh_audio->sample_format=AF_FORMAT_S32_LE; break;
-       }
-       break;
-    case 0x3: // IEEE float
-       sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
-       break;
-    case 0x6:  sh_audio->sample_format=AF_FORMAT_A_LAW;break;
-    case 0x7:  sh_audio->sample_format=AF_FORMAT_MU_LAW;break;
-    case 0x11: sh_audio->sample_format=AF_FORMAT_IMA_ADPCM;break;
-    case 0x50: sh_audio->sample_format=AF_FORMAT_MPEG2;break;
+    case 0x1:                  // Microsoft PCM
+    case 0xfffe:               // Extended
+        switch (sh_audio->samplesize) {
+        case 1: sh_audio->sample_format = AF_FORMAT_U8;     break;
+        case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
+        case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
+        case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
+        }
+        break;
+    case 0x3:                  // IEEE float
+        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+        break;
+    case 0x6:  sh_audio->sample_format = AF_FORMAT_A_LAW;      break;
+    case 0x7:  sh_audio->sample_format = AF_FORMAT_MU_LAW;     break;
+    case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM;  break;
+    case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2;      break;
 /*    case 0x2000: sh_audio->sample_format=AFMT_AC3; */
     case 0x20776172: // 'raw '
-       sh_audio->sample_format=AF_FORMAT_S16_BE;
-       if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_U8;
-       break;
+        sh_audio->sample_format = AF_FORMAT_S16_BE;
+        if (sh_audio->samplesize == 1)
+            sh_audio->sample_format = AF_FORMAT_U8;
+        break;
     case 0x736F7774: // 'twos'
-       sh_audio->sample_format=AF_FORMAT_S16_BE;
-       // intended fall-through
+        sh_audio->sample_format = AF_FORMAT_S16_BE;
+        // intended fall-through
     case 0x74776F73: // 'sowt'
-       if(sh_audio->samplesize==1) sh_audio->sample_format=AF_FORMAT_S8;
-       break;
+        if (sh_audio->samplesize == 1)
+            sh_audio->sample_format = AF_FORMAT_S8;
+        break;
     case 0x32336c66: // 'fl32', bigendian float32
-       sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
-       sh_audio->samplesize=4;
-       break;
+        sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
+        sh_audio->samplesize = 4;
+        break;
     case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
-       sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
-       sh_audio->samplesize=4;
-       break;
+        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+        sh_audio->samplesize = 4;
+        break;
 /*    case 0x34366c66: // 'fl64', bigendian float64
        sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
        sh_audio->samplesize=8;
@@ -98,93 +99,95 @@ static int init(sh_audio_t *sh_audio)
        sh_audio->samplesize=8;
        break;*/
     case 0x34326e69: // 'in24', bigendian int24
-       sh_audio->sample_format=AF_FORMAT_S24_BE;
-       sh_audio->samplesize=3;
-       break;
+        sh_audio->sample_format = AF_FORMAT_S24_BE;
+        sh_audio->samplesize = 3;
+        break;
     case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
-       sh_audio->sample_format=AF_FORMAT_S24_LE;
-       sh_audio->samplesize=3;
-       break;
+        sh_audio->sample_format = AF_FORMAT_S24_LE;
+        sh_audio->samplesize = 3;
+        break;
     case 0x32336e69: // 'in32', bigendian int32
-       sh_audio->sample_format=AF_FORMAT_S32_BE;
-       sh_audio->samplesize=4;
-       break;
+        sh_audio->sample_format = AF_FORMAT_S32_BE;
+        sh_audio->samplesize = 4;
+        break;
     case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
-       sh_audio->sample_format=AF_FORMAT_S32_LE;
-       sh_audio->samplesize=4;
-       break;
-    default: if(sh_audio->samplesize!=2) sh_audio->sample_format=AF_FORMAT_U8;
-  }
-  if (!sh_audio->samplesize) // this would cause MPlayer to hang later
-    sh_audio->samplesize = 2;
-  sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
-  return 1;
+        sh_audio->sample_format = AF_FORMAT_S32_LE;
+        sh_audio->samplesize = 4;
+        break;
+    default:
+        if (sh_audio->samplesize != 2)
+            sh_audio->sample_format = AF_FORMAT_U8;
+    }
+    if (!sh_audio->samplesize)  // this would cause MPlayer to hang later
+        sh_audio->samplesize = 2;
+    sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
+    return 1;
 }
 
-static int preinit(sh_audio_t *sh)
+static int preinit(sh_audio_t * sh)
 {
-  sh->audio_out_minsize=2048;
-  return 1;
+    sh->audio_out_minsize = 2048;
+    return 1;
 }
 
-static void uninit(sh_audio_t *sh)
+static void uninit(sh_audio_t * sh)
 {
     talloc_free(sh->context);
 }
 
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+static int control(sh_audio_t * sh, int cmd, void *arg, ...)
 {
-  int skip;
-    switch(cmd)
-    {
-      case ADCTRL_SKIP_FRAME:
-	skip=sh->i_bps/16;
-	skip=skip&(~3);
-	demux_read_data(sh->ds,NULL,skip);
-	return CONTROL_TRUE;
+    int skip;
+    switch (cmd) {
+    case ADCTRL_SKIP_FRAME:
+        skip = sh->i_bps / 16;
+        skip = skip & (~3);
+        demux_read_data(sh->ds, NULL, skip);
+        return CONTROL_TRUE;
     }
-  return CONTROL_UNKNOWN;
+    return CONTROL_UNKNOWN;
 }
 
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+static int decode_audio(sh_audio_t * sh_audio, unsigned char *buf, int minlen,
+                        int maxlen)
 {
-  unsigned len = sh_audio->channels*sh_audio->samplesize;
-  minlen = (minlen + len - 1) / len * len;
-  if (minlen > maxlen)
-      // if someone needs hundreds of channels adjust audio_out_minsize
-      // based on channels in preinit()
-      return -1;
+    unsigned len = sh_audio->channels * sh_audio->samplesize;
+    minlen = (minlen + len - 1) / len * len;
+    if (minlen > maxlen)
+        // if someone needs hundreds of channels adjust audio_out_minsize
+        // based on channels in preinit()
+        return -1;
 
-  len = 0;
-  struct ad_pcm_context *ctx = sh_audio->context;
-  while (len < minlen) {
-      if (ctx->packet_len == 0) {
-          double pts;
-          int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
-          if (plen < 0)
-              break;
-          ctx->packet_len = plen;
-          if (pts != MP_NOPTS_VALUE) {
-              sh_audio->pts = pts;
-              sh_audio->pts_bytes = 0;
-          }
-      }
-      int from_stored = ctx->packet_len;
-      if (from_stored > minlen - len)
-          from_stored = minlen - len;
-      memcpy(buf + len, ctx->packet_ptr, from_stored);
-      ctx->packet_len -= from_stored;
-      ctx->packet_ptr += from_stored;
-      sh_audio->pts_bytes += from_stored;
-      len += from_stored;
-  }
-  if (len == 0)
-      len = -1;  // The loop above only exits at error/EOF
-  if (len > 0 && sh_audio->channels >= 5) {
-    reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
-                        AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                        sh_audio->channels,
-                        len / sh_audio->samplesize, sh_audio->samplesize);
-  }
-  return len;
+    len = 0;
+    struct ad_pcm_context *ctx = sh_audio->context;
+    while (len < minlen) {
+        if (ctx->packet_len == 0) {
+            double pts;
+            int plen = ds_get_packet_pts(sh_audio->ds, &ctx->packet_ptr, &pts);
+            if (plen < 0)
+                break;
+            ctx->packet_len = plen;
+            if (pts != MP_NOPTS_VALUE) {
+                sh_audio->pts = pts;
+                sh_audio->pts_bytes = 0;
+            }
+        }
+        int from_stored = ctx->packet_len;
+        if (from_stored > minlen - len)
+            from_stored = minlen - len;
+        memcpy(buf + len, ctx->packet_ptr, from_stored);
+        ctx->packet_len -= from_stored;
+        ctx->packet_ptr += from_stored;
+        sh_audio->pts_bytes += from_stored;
+        len += from_stored;
+    }
+    if (len == 0)
+        len = -1;               // The loop above only exits at error/EOF
+    if (len > 0 && sh_audio->channels >= 5) {
+        reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
+                            AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+                            sh_audio->channels, len / sh_audio->samplesize,
+                            sh_audio->samplesize);
+    }
+    return len;
 }