make -srate work again, unify audio filter init and preinit.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@15812 b3059339-0415-0410-9bf9-f77b7e298cf2

libaf/af.c

@@ -329,12 +329,11 @@ void af_uninit(af_stream_t* s)
    and output should contain the format of the current movie and the
    formate of the preferred output respectively. The function is
    reentrant i.e. if called with an already initialized stream the
-   stream will be reinitialized. If the binary parameter
-   "force_output" is set, the output format will be converted to the
-   format given in "s", otherwise the output fromat in the last filter
-   will be copied "s". The return value is 0 if success and -1 if
-   failure */
-int af_init(af_stream_t* s, int force_output)
+   stream will be reinitialized.
+   If one of the prefered output parameters is 0 the one that needs
+   no conversion is used (i.e. the output format in the last filter).
+   The return value is 0 if success and -1 if failure */
+int af_init(af_stream_t* s)
 {
   int i=0;
 
@@ -373,17 +372,11 @@ int af_init(af_stream_t* s, int force_output)
     if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first))
       return -1;
 
-  // If force_output isn't set do not compensate for output format
-  if(!force_output){
-    memcpy(&s->output, s->last->data, sizeof(af_data_t));
-    return 0;
-  }
-
   // Check output format
   if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
     af_instance_t* af = NULL; // New filter
     // Check output frequency if not OK fix with resample
-    if(s->last->data->rate!=s->output.rate){
+    if(s->output.rate && s->last->data->rate!=s->output.rate){
       // try to find a filter that can change samplrate
       af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
                &(s->output.rate));
@@ -428,7 +421,7 @@ int af_init(af_stream_t* s, int force_output)
       
     // Check number of output channels fix if not OK
     // If needed always inserted last -> easy to screw up other filters
-    if(s->last->data->nch!=s->output.nch){
+    if(s->output.nch && s->last->data->nch!=s->output.nch){
       if(!strcmp(s->last->info->name,"format"))
 	af = af_prepend(s,s->last,"channels");
       else
@@ -441,7 +434,7 @@ 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){
+    if(s->output.format && s->last->data->format != s->output.format){
       if(strcmp(s->last->info->name,"format"))
 	af = af_append(s,s->last,"format");
       else
@@ -458,6 +451,9 @@ int af_init(af_stream_t* s, int force_output)
     if(AF_OK != af_reinit(s,s->first))
       return -1;
 
+    if (!s->output.format) s->output.format = s->last->data->format;
+    if (!s->output.nch) s->output.nch = s->last->data->nch;
+    if (!s->output.rate) s->output.rate = s->last->data->rate;
     if((s->last->data->format != s->output.format) || 
        (s->last->data->nch    != s->output.nch)    || 
        (s->last->data->rate   != s->output.rate))  {

libaf/af.h

@@ -134,7 +134,7 @@ typedef struct af_stream_s
    format given in "s", otherwise the output fromat in the last filter
    will be copied "s". The return value is 0 if success and -1 if
    failure */
-int af_init(af_stream_t* s, int force_output);
+int af_init(af_stream_t* s);
 
 // Uninit and remove all filters
 void af_uninit(af_stream_t* s);

libmpcodecs/dec_audio.c

@@ -260,6 +260,7 @@ void uninit_audio(sh_audio_t *sh_audio)
     }
     if(sh_audio->a_out_buffer!=sh_audio->a_buffer) free(sh_audio->a_out_buffer);
     sh_audio->a_out_buffer=NULL;
+    sh_audio->a_out_buffer_size=0;
     if(sh_audio->a_buffer) free(sh_audio->a_buffer);
     sh_audio->a_buffer=NULL;
     if(sh_audio->a_in_buffer) free(sh_audio->a_in_buffer);
@@ -270,50 +271,14 @@ void uninit_audio(sh_audio_t *sh_audio)
 int preinit_audio_filters(sh_audio_t *sh_audio, 
 	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 = in_format;
-  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 ? *out_format : afs->input.format;
-  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/%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)){
-    free(afs);
-    return 0; // failed :(
-  }
-  
-  *out_samplerate=afs->output.rate;
-  *out_channels=afs->output.nch;
-  *out_format=afs->output.format;
-
-  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;
+  init_audio_filters(sh_audio, in_samplerate, in_channels, in_format,
+      out_samplerate, out_channels, out_format, 0, 0);
 }
 
  /* Init audio filters */
 int init_audio_filters(sh_audio_t *sh_audio, 
 	int in_samplerate, int in_channels, int in_format,
-	int out_samplerate, int out_channels, int out_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){
@@ -328,9 +293,9 @@ int init_audio_filters(sh_audio_t *sh_audio,
   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 ? out_format : afs->input.format;
+  afs->output.rate   = *out_samplerate;
+  afs->output.nch    = *out_channels;
+  afs->output.format = *out_format;
   af_fix_parameters(&(afs->output));
 
   // filter config:  
@@ -341,12 +306,17 @@ int init_audio_filters(sh_audio_t *sh_audio,
       afs->output.rate,afs->output.nch,af_fmt2str_short(afs->output.format));
   
   // let's autoprobe it!
-  if(0 != af_init(afs,1)){
+  if(0 != af_init(afs)){
     sh_audio->afilter=NULL;
     free(afs);
     return 0; // failed :(
   }
+
+  *out_samplerate=afs->output.rate;
+  *out_channels=afs->output.nch;
+  *out_format=afs->output.format;
   
+  if (out_maxsize || out_minsize) {
   // allocate the a_out_* buffers:
   if(out_maxsize<out_minsize) out_maxsize=out_minsize;
   if(out_maxsize<8192) out_maxsize=MAX_OUTBURST; // not sure this is ok
@@ -355,6 +325,7 @@ int init_audio_filters(sh_audio_t *sh_audio,
   sh_audio->a_out_buffer=malloc(sh_audio->a_out_buffer_size);
   memset(sh_audio->a_out_buffer,0,sh_audio->a_out_buffer_size);
   sh_audio->a_out_buffer_len=0;
+  }
   
   // ok!
   sh_audio->afilter=(void*)afs;

libmpcodecs/dec_audio.h

@@ -12,7 +12,7 @@ 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 out_samplerate, int out_channels, int out_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,

mencoder.c

@@ -819,9 +819,9 @@ mux_a->source=sh_audio;
 
 mux_a->codec=out_audio_codec;
 
-ao_data.samplerate = force_srate ? force_srate : new_srate;
-ao_data.channels = audio_output_channels ? audio_output_channels : sh_audio->channels;
-ao_data.format = audio_output_format ? audio_output_format : sh_audio->sample_format;
+ao_data.samplerate = force_srate;
+ao_data.channels = 0;
+ao_data.format = audio_output_format;
 if(!preinit_audio_filters(sh_audio,
    // input:
    new_srate,
@@ -833,7 +833,7 @@ if(!preinit_audio_filters(sh_audio,
    }
 
 aparams.channels = ao_data.channels;
-aparams.sample_rate = force_srate ? force_srate : ao_data.samplerate;
+aparams.sample_rate = ao_data.samplerate;
 aparams.audio_preload = 1000 * audio_preload;
 if(mux_a->codec != ACODEC_COPY) {
     aencoder = new_audio_encoder(mux_a, &aparams);
@@ -841,7 +841,7 @@ if(mux_a->codec != ACODEC_COPY) {
         mencoder_exit(1, NULL);
     if(!init_audio_filters(sh_audio, 
         new_srate, sh_audio->channels, sh_audio->sample_format,  
-        aparams.sample_rate, aparams.channels, aencoder->input_format, 
+        &aparams.sample_rate, &aparams.channels, &aencoder->input_format, 
         aencoder->min_buffer_size, aencoder->max_buffer_size)) {
       mp_msg(MSGT_CPLAYER,MSGL_FATAL,MSGTR_NoMatchingFilter);
       mencoder_exit(1,NULL);
@@ -951,9 +951,9 @@ else if (sh_audio) {
 	    new_srate,
 	    sh_audio->channels,
 	    sh_audio->sample_format,
-	    mux_a->wf->nSamplesPerSec,
-	    mux_a->wf->nChannels,
-	    out_format,
+	    &mux_a->wf->nSamplesPerSec,
+	    &mux_a->wf->nChannels,
+	    &out_format,
 	    out_minsize,
 	    out_maxsize))
 	{

mplayer.c

@@ -955,7 +955,7 @@ static int build_afilter_chain(sh_audio_t *sh_audio, ao_data_t *ao_data)
   }
   result =  init_audio_filters(sh_audio, new_srate,
            sh_audio->channels, sh_audio->sample_format,
-           ao_data->samplerate, ao_data->channels, ao_data->format,
+           &ao_data->samplerate, &ao_data->channels, &ao_data->format,
            ao_data->outburst * 4, ao_data->buffersize);
   mixer.afilter = sh_audio->afilter;
 #ifdef HAVE_NEW_GUI
@@ -2133,10 +2133,11 @@ osd_text_buffer[0]=0;
 if(sh_audio){
   //const ao_info_t *info=audio_out->info;
   current_module="af_preinit";
-  ao_data.samplerate=force_srate?force_srate:sh_audio->samplerate*playback_speed;
-  ao_data.channels=audio_output_channels?audio_output_channels:sh_audio->channels;
-  ao_data.format=audio_output_format?audio_output_format:sh_audio->sample_format;
+  ao_data.samplerate=force_srate;
+  ao_data.channels=0;
+  ao_data.format=audio_output_format;
 #if 1
+  // first init to detect best values
   if(!preinit_audio_filters(sh_audio,
         // input:
         (int)(sh_audio->samplerate*playback_speed),