audio/filter: remove redundant log message prefixes

These are now appended automatically, so you'd get them twice before
this commit.

13 files changed, 44 insertions, 44 deletions

diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index 1ea31b260a..08d037bf4e 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -142,12 +142,12 @@ static int control(struct af_instance *af, int cmd, void *arg)
         // bs2b have srate limits, try to resample if needed
         if (af->data->rate > BS2B_MAXSRATE || af->data->rate < BS2B_MINSRATE) {
             af->data->rate = BS2B_DEFAULT_SRATE;
-            MP_WARN(af, "[bs2b] Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
-                   "[bs2b] Trying to resample to %d Hz.\n",
+            MP_WARN(af, "Requested sample rate %d Hz is out of bounds [%d..%d] Hz.\n"
+                   "Trying to resample to %d Hz.\n",
                    af->data->rate, BS2B_MINSRATE, BS2B_MAXSRATE, BS2B_DEFAULT_SRATE);
         }
         bs2b_set_srate(s->filter, (long)af->data->rate);
-        MP_VERBOSE(af, "[bs2b] using format %s\n",
+        MP_VERBOSE(af, "using format %s\n",
                af_fmt_to_str(af->data->format));
 
         return af_test_output(af,(struct mp_audio*)arg);
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index 74a0de0fd6..6db44ba024 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -124,14 +124,14 @@ static int check_routes(struct af_instance *af, int nin, int nout)
   af_channels_t* s = af->priv;
   int i;
   if((s->nr < 1) || (s->nr > AF_NCH)){
-    MP_ERR(af, "[channels] The number of routing pairs must be"
+    MP_ERR(af, "The number of routing pairs must be"
 	   " between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
     return AF_ERROR;
   }
 
   for(i=0;i<s->nr;i++){
     if((s->route[i][FR] >= nin) || (s->route[i][TO] >= nout)){
-      MP_ERR(af, "[channels] Invalid routing in pair nr. %i.\n", i);
+      MP_ERR(af, "Invalid routing in pair nr. %i.\n", i);
       return AF_ERROR;
     }
   }
@@ -220,11 +220,11 @@ static int af_open(struct af_instance* af){
         do {
             int n = 0;
             if (ch >= AF_NCH) {
-                MP_FATAL(af, "[channels] Can't have more than %d routes.\n", AF_NCH);
+                MP_FATAL(af, "Can't have more than %d routes.\n", AF_NCH);
                 return AF_ERROR;
             }
             sscanf(cp, "%i-%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n);
-            MP_VERBOSE(af, "[channels] Routing from channel %i to"
+            MP_VERBOSE(af, "Routing from channel %i to"
                 " channel %i\n",s->route[ch][FR],s->route[ch][TO]);
             cp = &cp[n];
             ch++;
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index 51c453f6e5..a8cd79f117 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -54,7 +54,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     struct mp_audio *in = arg;
 
     if (in->bps != 1 && in->bps != 2 && in->bps != 4) {
-      MP_FATAL(af, "[delay] Sample format not supported\n");
+      MP_FATAL(af, "Sample format not supported\n");
       return AF_ERROR;
     }
 
@@ -69,16 +69,16 @@ static int control(struct af_instance* af, int cmd, void* arg)
     for(i=0;i<af->data->nch;i++){
       s->q[i] = calloc(L,af->data->bps);
       if(NULL == s->q[i])
-	MP_FATAL(af, "[delay] Out of memory\n");
+	MP_FATAL(af, "Out of memory\n");
     }
 
     if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0))
       return AF_ERROR;
     s->ri = 0;
     for(i=0;i<AF_NCH;i++){
-      MP_DBG(af, "[delay] Channel %i delayed by %0.3fms\n",
+      MP_DBG(af, "Channel %i delayed by %0.3fms\n",
              i,MPCLAMP(s->d[i],0.0,1000.0));
-      MP_TRACE(af, "[delay] Channel %i delayed by %i samples\n",
+      MP_TRACE(af, "Channel %i delayed by %i samples\n",
              i,s->wi[i]);
     }
     return AF_OK;
diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c
index 6822523efe..c13c32b968 100644
--- a/audio/filter/af_dummy.c
+++ b/audio/filter/af_dummy.c
@@ -33,7 +33,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
   switch(cmd){
   case AF_CONTROL_REINIT: ;
     *af->data = *(struct mp_audio*)arg;
-    MP_VERBOSE(af, "[dummy] Was reinitialized: %iHz/%ich/%s\n",
+    MP_VERBOSE(af, "Was reinitialized: %iHz/%ich/%s\n",
 	af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
     return AF_OK;
   }
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index 49aedeb106..4f5a29706e 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -107,7 +107,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
       s->K--;
 
     if(s->K != KM)
-      MP_INFO(af, "[equalizer] Limiting the number of filters to"
+      MP_INFO(af, "Limiting the number of filters to"
 	     " %i due to low sample rate.\n",s->K);
 
     // Generate filter taps
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index a5a8257c39..542b2a1f9f 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -95,20 +95,20 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Allocate new buffers (as one continuous block)
     s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
     if(NULL == s->buf[0])
-      MP_FATAL(af, "[export] Out of memory\n");
+      MP_FATAL(af, "Out of memory\n");
     for(i = 1; i < af->data->nch; i++)
       s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps;
 
     if (!s->filename) {
-        MP_FATAL(af, "[export] No filename set.\n");
+        MP_FATAL(af, "No filename set.\n");
         return AF_ERROR;
     }
 
     // Init memory mapping
     s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640);
-    MP_INFO(af, "[export] Exporting to file: %s\n", s->filename);
+    MP_INFO(af, "Exporting to file: %s\n", s->filename);
     if(s->fd < 0) {
-      MP_FATAL(af, "[export] Could not open/create file: %s\n",
+      MP_FATAL(af, "Could not open/create file: %s\n",
 	     s->filename);
       return AF_ERROR;
     }
@@ -125,8 +125,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // mmap size
     s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0);
     if(s->mmap_area == NULL)
-      MP_FATAL(af, "[export] Could not mmap file %s\n", s->filename);
-    MP_INFO(af, "[export] Memory mapped to file: %s (%p)\n",
+      MP_FATAL(af, "Could not mmap file %s\n", s->filename);
+    MP_INFO(af, "Memory mapped to file: %s (%p)\n",
 	   s->filename, s->mmap_area);
 
     // Initialize header
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index e3fecb8ad7..e329b4e558 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -296,7 +296,7 @@ static int control(struct af_instance *af, int cmd, void* arg)
 	if(af->data->rate != 48000) {
 	    // automatic samplerate adjustment in the filter chain
 	    // is not yet supported.
-	    MP_ERR(af, "[hrtf] ERROR: Sampling rate is not 48000 Hz (%d)!\n",
+	    MP_ERR(af, "ERROR: Sampling rate is not 48000 Hz (%d)!\n",
 		   af->data->rate);
 	    return AF_ERROR;
 	}
@@ -367,25 +367,25 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags)
 	s->print_flag = 0;
 	switch (s->decode_mode) {
 	case HRTF_MIX_51:
-	  MP_INFO(af, "[hrtf] Using HRTF to mix %s discrete surround into "
+	  MP_INFO(af, "Using HRTF to mix %s discrete surround into "
 		 "L, R channels\n", s->matrix_mode ? "5+1" : "5");
 	  break;
 	case HRTF_MIX_STEREO:
-	  MP_INFO(af, "[hrtf] Using HRTF to mix stereo into "
+	  MP_INFO(af, "Using HRTF to mix stereo into "
 		 "L, R channels\n");
 	  break;
 	case HRTF_MIX_MATRIX2CH:
-	  MP_INFO(af, "[hrtf] Using active matrix to decode 2 channel "
+	  MP_INFO(af, "Using active matrix to decode 2 channel "
 		 "input, HRTF to mix %s matrix surround into "
 		 "L, R channels\n", "3/2");
 	  break;
 	default:
-	  MP_WARN(af, "[hrtf] bogus decode_mode: %d\n", s->decode_mode);
+	  MP_WARN(af, "bogus decode_mode: %d\n", s->decode_mode);
 	  break;
 	}
 
        if(s->matrix_mode)
-	  MP_INFO(af, "[hrtf] Using active matrix to decode rear center "
+	  MP_INFO(af, "Using active matrix to decode rear center "
 		 "channel\n");
     }
 
@@ -595,7 +595,7 @@ static int af_open(struct af_instance* af)
     s->print_flag = 1;
 
     if (allocate(s) != 0) {
- 	MP_ERR(af, "[hrtf] Memory allocation error.\n");
+ 	MP_ERR(af, "Memory allocation error.\n");
 	return AF_ERROR;
     }
 
@@ -614,13 +614,13 @@ static int af_open(struct af_instance* af)
     s->cr_ir = cr_filt + (s->cr_o = pulse_detect(cr_filt));
 
     if((s->ba_ir = malloc(s->basslen * sizeof(float))) == NULL) {
- 	MP_ERR(af, "[hrtf] Memory allocation error.\n");
+ 	MP_ERR(af, "Memory allocation error.\n");
 	return AF_ERROR;
     }
     fc = 2.0 * BASSFILTFREQ / (float)af->data->rate;
     if(af_filter_design_fir(s->basslen, s->ba_ir, &fc, LP | KAISER, 4 * M_PI) ==
        -1) {
-	MP_ERR(af, "[hrtf] Unable to design low-pass "
+	MP_ERR(af, "Unable to design low-pass "
 	       "filter.\n");
 	return AF_ERROR;
     }
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 6df3820e75..34de6adb74 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -187,7 +187,7 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
 
         AVFrame *frame = avcodec_alloc_frame();
         if (!frame) {
-            MP_FATAL(af, "[libaf] Could not allocate memory \n");
+            MP_FATAL(af, "Could not allocate memory \n");
             return -1;
         }
         frame->nb_samples = s->in_samples;
@@ -201,7 +201,7 @@ static int filter(struct af_instance* af, struct mp_audio* audio, int flags)
         int ok;
         ret = avcodec_encode_audio2(s->lavc_actx, &s->pkt, frame, &ok);
         if (ret < 0 || !ok) {
-            MP_FATAL(af, "[lavac3enc] Encode failed.\n");
+            MP_FATAL(af, "Encode failed.\n");
             return -1;
         }
 
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index 4bd95459c0..f6537f1db0 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -221,7 +221,7 @@ static int configure_lavrr(struct af_instance *af, struct mp_audio *in,
     if (avresample_open(s->avrctx) < 0 ||
         avresample_open(s->avrctx_out) < 0)
     {
-        MP_ERR(af, "[lavrresample] Cannot open "
+        MP_ERR(af, "Cannot open "
                 "Libavresample Context. \n");
         return AF_ERROR;
     }
@@ -397,7 +397,7 @@ static int af_open(struct af_instance *af)
     if (s->avrctx && s->avrctx_out) {
         return AF_OK;
     } else {
-        MP_ERR(af, "[lavrresample] Cannot initialize "
+        MP_ERR(af, "Cannot initialize "
                "Libavresample Context. \n");
         uninit(af);
         return AF_ERROR;
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index 081a8f72d0..01b0575ae3 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -80,7 +80,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     // Sanity check
     if(((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH){
-      MP_ERR(af, "[pan] The number of output channels must be"
+      MP_ERR(af, "The number of output channels must be"
             " between 1 and %i. Current value is %i\n",AF_NCH,((int*)arg)[0]);
       return AF_ERROR;
     }
@@ -161,7 +161,7 @@ static int af_open(struct af_instance* af){
     j = 0; k = 0;
     while(cp && k < AF_NCH){
         sscanf(cp, "%f%n" , &s->level[j][k], &n);
-        MP_VERBOSE(af, "[pan] Pan level from channel %i to"
+        MP_VERBOSE(af, "Pan level from channel %i to"
                 " channel %i = %f\n",k,j,s->level[j][k]);
         cp =&cp[n];
         j++;
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index eb5fc66a12..cd288e9d12 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -275,7 +275,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         int nch = data->nch;
         int use_int = 0;
 
-        MP_VERBOSE(af, "[scaletempo] %.3f speed * %.3f scale_nominal = %.3f\n",
+        MP_VERBOSE(af, "%.3f speed * %.3f scale_nominal = %.3f\n",
                s->speed, s->scale_nominal, s->scale);
 
         mp_audio_force_interleaved_format(data);
@@ -317,7 +317,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
             s->buf_overlap      = realloc(s->buf_overlap, s->bytes_overlap);
             s->table_blend      = realloc(s->table_blend, s->bytes_overlap * 4);
             if (!s->buf_overlap || !s->table_blend) {
-                MP_FATAL(af, "[scaletempo] Out of memory\n");
+                MP_FATAL(af, "Out of memory\n");
                 return AF_ERROR;
             }
             memset(s->buf_overlap, 0, s->bytes_overlap);
@@ -354,7 +354,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 s->table_window = realloc(s->table_window,
                                           s->bytes_overlap * 2 - nch * bps * 2);
                 if (!s->buf_pre_corr || !s->table_window) {
-                    MP_FATAL(af, "[scaletempo] Out of memory\n");
+                    MP_FATAL(af, "Out of memory\n");
                     return AF_ERROR;
                 }
                 memset((char *)s->buf_pre_corr + s->bytes_overlap * 2, 0,
@@ -371,7 +371,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 s->table_window = realloc(s->table_window,
                                           s->bytes_overlap - nch * bps);
                 if (!s->buf_pre_corr || !s->table_window) {
-                    MP_FATAL(af, "[scaletempo] Out of memory\n");
+                    MP_FATAL(af, "Out of memory\n");
                     return AF_ERROR;
                 }
                 float *pw = s->table_window;
@@ -391,14 +391,14 @@ static int control(struct af_instance *af, int cmd, void *arg)
                          * bps * nch;
         s->buf_queue = realloc(s->buf_queue, s->bytes_queue + UNROLL_PADDING);
         if (!s->buf_queue) {
-            MP_FATAL(af, "[scaletempo] Out of memory\n");
+            MP_FATAL(af, "Out of memory\n");
             return AF_ERROR;
         }
 
         s->bytes_queued = 0;
         s->bytes_to_slide = 0;
 
-        MP_DBG(af, "[scaletempo] "
+        MP_DBG(af, ""
                "%.2f stride_in, %i stride_out, %i standing, "
                "%i overlap, %i search, %i queue, %s mode\n",
                s->frames_stride_scaled,
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index 37952ab0be..9719d46828 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -99,7 +99,7 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
     s->pos += 2 * M_PI * s->freq / data->rate;
   }
 
-   MP_VERBOSE(af, "[sinesuppress] f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
+   MP_VERBOSE(af, "f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
 
   return 0;
 }
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c
index daa1f25ef1..9cabeedf9d 100644
--- a/audio/filter/af_surround.c
+++ b/audio/filter/af_surround.c
@@ -94,7 +94,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     struct mp_audio *in = arg;
     float fc;
     if (!mp_chmap_is_stereo(&in->channels)) {
-        MP_ERR(af, "[surround] Only stereo input is supported.\n");
+        MP_ERR(af, "Only stereo input is supported.\n");
         return AF_DETACH;
     }
 
@@ -105,7 +105,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Surround filer coefficients
     fc = 2.0 * 7000.0/(float)af->data->rate;
     if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
-      MP_ERR(af, "[surround] Unable to design low-pass filter.\n");
+      MP_ERR(af, "Unable to design low-pass filter.\n");
       return AF_ERROR;
     }
 
@@ -116,7 +116,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     s->dl = calloc(LD,af->data->bps);
     s->dr = calloc(LD,af->data->bps);
     if((NULL == s->dl) || (NULL == s->dr))
-      MP_FATAL(af, "[delay] Out of memory\n");
+      MP_FATAL(af, "Out of memory\n");
 
     // Initialize delay queue index
     if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))