finally remove the refences to bps outside libaf. also simplification of some messages and removed redundants

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14819 b3059339-0415-0410-9bf9-f77b7e298cf2

5 files changed, 41 insertions, 49 deletions

diff --git a/libaf/af.c b/libaf/af.c
index cbafe3a5e6..b14f578860 100644
--- a/libaf/af.c
+++ b/libaf/af.c
@@ -265,8 +265,7 @@ int af_reinit(af_stream_t* s, af_instance_t* af)
 	    return rv;
 	}
 	// Insert format filter
-	if(((af->prev?af->prev->data->format:s->input.format) != in.format) || 
-	   ((af->prev?af->prev->data->bps:s->input.bps) != in.bps)){
+	if((af->prev?af->prev->data->format:s->input.format) != in.format){
 	  // Create format filter
 	  if(NULL == (new = af_prepend(s,af,"format")))
 	    return AF_ERROR;
@@ -438,8 +437,7 @@ int af_init(af_stream_t* s, int force_output)
     }
     
     // Check output format fix if not OK
-    if((s->last->data->format != s->output.format) || 
-       (s->last->data->bps != s->output.bps)){
+    if(s->last->data->format != s->output.format){
       if(strcmp(s->last->info->name,"format"))
 	af = af_append(s,s->last,"format");
       else
@@ -457,7 +455,6 @@ int af_init(af_stream_t* s, int force_output)
       return -1;
 
     if((s->last->data->format != s->output.format) || 
-       (s->last->data->bps    != s->output.bps)    ||
        (s->last->data->nch    != s->output.nch)    || 
        (s->last->data->rate   != s->output.rate))  {
       // Something is stuffed audio out will not work 
@@ -698,3 +695,7 @@ void af_help (void) {
   }
 }
 
+void af_fix_parameters(af_data_t *data)
+{
+    data->bps = af_fmt2bits(data->format)/8;
+}
diff --git a/libaf/af.h b/libaf/af.h
index 03c825a20e..d4ff9d1c30 100644
--- a/libaf/af.h
+++ b/libaf/af.h
@@ -214,6 +214,10 @@ float af_softclip(float a);
 /** Print a list of all available audio filters */
 void af_help(void);
 
+/* Fill the missing parameters in the af_data_t structure.
+   Used for stuffing bps with a value based on format. */
+void af_fix_paramaters(af_data_t *data);
+
 /* Memory reallocation macro: if a local buffer is used (i.e. if the
    filter doesn't operate on the incoming buffer this macro must be
    called to ensure the buffer is big enough. */
@@ -269,5 +273,3 @@ extern af_msg_cfg_t af_msg_cfg; // Message
 #endif
 
 #endif /* __aop_h__ */
-
-
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index ad09b1c3ab..5a01b318f1 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -265,32 +265,29 @@ void uninit_audio(sh_audio_t *sh_audio)
 
  /* Init audio filters */
 int preinit_audio_filters(sh_audio_t *sh_audio, 
-	int in_samplerate, int in_channels, int in_format, int in_bps,
-	int* out_samplerate, int* out_channels, int* out_format, int out_bps){
-  char strbuf[200];
+	int in_samplerate, int in_channels, int in_format,
+	int* out_samplerate, int* out_channels, int* out_format){
   af_stream_t* afs=malloc(sizeof(af_stream_t));
   memset(afs,0,sizeof(af_stream_t));
 
   // input format: same as codec's output format:
   afs->input.rate   = in_samplerate;
   afs->input.nch    = in_channels;
-//  afs->input.format = af_format_decode(in_format);
   afs->input.format = in_format;
-  afs->input.bps    = in_bps;
+  af_fix_parameters(&(afs->input));
 
   // output format: same as ao driver's input format (if missing, fallback to input)
   afs->output.rate   = *out_samplerate ? *out_samplerate : afs->input.rate;
   afs->output.nch    = *out_channels ? *out_channels : afs->input.nch;
-//  afs->output.format = *out_format ? af_format_decode(*out_format) : afs->input.format;
   afs->output.format = *out_format ? *out_format : afs->input.format;
-  afs->output.bps    = out_bps ? out_bps : afs->input.bps;
+  af_fix_parameters(&(afs->output));
 
   // filter config:  
   memcpy(&afs->cfg,&af_cfg,sizeof(af_cfg_t));
   
-  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Checking audio filter chain for %dHz/%dch/%dbit -> %dHz/%dch/%dbit...\n",
-      afs->input.rate,afs->input.nch,afs->input.bps*8,
-      afs->output.rate,afs->output.nch,afs->output.bps*8);
+  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Checking audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
+      afs->input.rate,afs->input.nch,af_fmt2str_short(afs->input.format),
+      afs->output.rate,afs->output.nch,af_fmt2str_short(afs->output.format));
   
   // let's autoprobe it!
   if(0 != af_init(afs,0)){
@@ -300,12 +297,11 @@ int preinit_audio_filters(sh_audio_t *sh_audio,
   
   *out_samplerate=afs->output.rate;
   *out_channels=afs->output.nch;
-//  *out_format=af_format_encode((void*)(&afs->output));
   *out_format=afs->output.format;
-  
-  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "AF_pre: af format: %d bps, %d ch, %d hz, %s\n",
-      afs->output.bps, afs->output.nch, afs->output.rate,
-      af_fmt2str(afs->output.format,strbuf,200));
+
+  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "AF_pre: %dHz/%dch/%s\n",
+      afs->output.rate, afs->output.nch,
+      af_fmt2str_short(afs->output.format));
   
   sh_audio->afilter=(void*)afs;
   return 1;
@@ -313,8 +309,8 @@ int preinit_audio_filters(sh_audio_t *sh_audio,
 
  /* Init audio filters */
 int init_audio_filters(sh_audio_t *sh_audio, 
-	int in_samplerate, int in_channels, int in_format, int in_bps,
-	int out_samplerate, int out_channels, int out_format, int out_bps,
+	int in_samplerate, int in_channels, int in_format,
+	int out_samplerate, int out_channels, int out_format,
 	int out_minsize, int out_maxsize){
   af_stream_t* afs=sh_audio->afilter;
   if(!afs){
@@ -325,23 +321,21 @@ int init_audio_filters(sh_audio_t *sh_audio,
   // input format: same as codec's output format:
   afs->input.rate   = in_samplerate;
   afs->input.nch    = in_channels;
-//  afs->input.format = af_format_decode(in_format);
   afs->input.format = in_format;
-  afs->input.bps    = in_bps;
+  af_fix_parameters(&(afs->input));
 
   // output format: same as ao driver's input format (if missing, fallback to input)
   afs->output.rate   = out_samplerate ? out_samplerate : afs->input.rate;
   afs->output.nch    = out_channels ? out_channels : afs->input.nch;
-//  afs->output.format = af_format_decode(out_format ? out_format : afs->input.format);
   afs->output.format = out_format ? out_format : afs->input.format;
-  afs->output.bps    = out_bps ? out_bps : afs->input.bps;
+  af_fix_parameters(&(afs->output));
 
   // filter config:  
   memcpy(&afs->cfg,&af_cfg,sizeof(af_cfg_t));
   
-  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Building audio filter chain for %dHz/%dch/%dbit -> %dHz/%dch/%dbit...\n",
-      afs->input.rate,afs->input.nch,afs->input.bps*8,
-      afs->output.rate,afs->output.nch,afs->output.bps*8);
+  mp_msg(MSGT_DECAUDIO, MSGL_INFO, "Building audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
+      afs->input.rate,afs->input.nch,af_fmt2str_short(afs->input.format),
+      afs->output.rate,afs->output.nch,af_fmt2str_short(afs->output.format));
   
   // let's autoprobe it!
   if(0 != af_init(afs,1)){
@@ -416,9 +410,8 @@ int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
   afd.len=declen;
   afd.rate=sh_audio->samplerate;
   afd.nch=sh_audio->channels;
-//  afd.format=af_format_decode(sh_audio->sample_format);
   afd.format=sh_audio->sample_format;
-  afd.bps=sh_audio->samplesize;
+  af_fix_parameters(&afd);
   //pafd=&afd;
 //  printf("\nAF: %d --> ",declen);
   pafd=af_play(sh_audio->afilter,&afd);
@@ -460,3 +453,7 @@ void skip_audio_frame(sh_audio_t *sh_audio)
   // default skip code:
   ds_fill_buffer(sh_audio->ds);  // skip block
 }
+
+void adjust_volume()
+{
+}
diff --git a/libmpcodecs/dec_audio.h b/libmpcodecs/dec_audio.h
index db4ceebc24..ad8872737e 100644
--- a/libmpcodecs/dec_audio.h
+++ b/libmpcodecs/dec_audio.h
@@ -11,9 +11,9 @@ extern void skip_audio_frame(sh_audio_t *sh_audio);
 extern void uninit_audio(sh_audio_t *sh_audio);
 
 extern int init_audio_filters(sh_audio_t *sh_audio, 
-	int in_samplerate, int in_channels, int in_format, int in_bps,
-	int out_samplerate, int out_channels, int out_format, int out_bps,
+	int in_samplerate, int in_channels, int in_format,
+	int out_samplerate, int out_channels, int out_format,
 	int out_minsize, int out_maxsize);
 extern int preinit_audio_filters(sh_audio_t *sh_audio,
-        int in_samplerate, int in_channels, int in_format, int in_bps,
-        int* out_samplerate, int* out_channels, int* out_format, int out_bps);
+        int in_samplerate, int in_channels, int in_format,
+        int* out_samplerate, int* out_channels, int* out_format);
diff --git a/mplayer.c b/mplayer.c
index 73d350e4e6..19cdd0daa8 100644
--- a/mplayer.c
+++ b/mplayer.c
@@ -950,9 +950,8 @@ static int build_afilter_chain(sh_audio_t *sh_audio, ao_data_t *ao_data)
     playback_speed = (float)new_srate / (float)sh_audio->samplerate;
   }
   result =  init_audio_filters(sh_audio, new_srate,
-           sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize,
+           sh_audio->channels, sh_audio->sample_format,
            ao_data->samplerate, ao_data->channels, ao_data->format,
-           af_fmt2bits(ao_data->format) / 8, /* ao_data.bps, */
            ao_data->outburst * 4, ao_data->buffersize);
   mixer.afilter = sh_audio->afilter;
 #ifdef HAVE_NEW_GUI
@@ -2119,16 +2118,10 @@ if(sh_audio){
   if(!preinit_audio_filters(sh_audio,
         // input:
         (int)(sh_audio->samplerate*playback_speed),
-	sh_audio->channels, sh_audio->sample_format, sh_audio->samplesize,
+	sh_audio->channels, sh_audio->sample_format,
 	// output:
-	&ao_data.samplerate, &ao_data.channels, &ao_data.format,
-	af_fmt2bits(ao_data.format)/8)){
+	&ao_data.samplerate, &ao_data.channels, &ao_data.format)){
       mp_msg(MSGT_CPLAYER,MSGL_ERR,MSGTR_AudioFilterChainPreinitError);
-  } else {
-    char buf[128];
-    mp_msg(MSGT_CPLAYER,MSGL_INFO,"AF_pre: %dHz %dch %s\n",
-      ao_data.samplerate, ao_data.channels,
-      af_fmt2str(ao_data.format, buf, 128));
   }
 #endif  
   current_module="ao2_init";
@@ -2143,12 +2136,11 @@ if(sh_audio){
     sh_audio=d_audio->sh=NULL; // -> nosound
   } else {
     // SUCCESS:
-    char buf[128];
     inited_flags|=INITED_AO;
     mp_msg(MSGT_CPLAYER,MSGL_INFO,"AO: [%s] %dHz %dch %s (%d bps)\n",
       audio_out->info->short_name,
       ao_data.samplerate, ao_data.channels,
-      af_fmt2str(ao_data.format, buf, 128),
+      af_fmt2str_short(ao_data.format),
       af_fmt2bits(ao_data.format)/8 );
     mp_msg(MSGT_CPLAYER,MSGL_V,"AO: Description: %s\nAO: Author: %s\n",
       audio_out->info->name, audio_out->info->author);