audio: drop "_NE"/"ne" suffix from audio formats

You get the native format by not appending any suffix to the format.

This change includes user-facing names, e.g. for the --format option.

27 files changed, 56 insertions, 68 deletions

diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
index 322f45826f..777c20c2c9 100644
--- a/audio/decode/ad_mpg123.c
+++ b/audio/decode/ad_mpg123.c
@@ -149,13 +149,13 @@ static int set_format(sh_audio_t *sh)
             sh->sample_format = AF_FORMAT_S8;
             break;
         case MPG123_ENC_SIGNED_16:
-            sh->sample_format = AF_FORMAT_S16_NE;
+            sh->sample_format = AF_FORMAT_S16;
             break;
         case MPG123_ENC_SIGNED_32:
-            sh->sample_format = AF_FORMAT_S32_NE;
+            sh->sample_format = AF_FORMAT_S32;
             break;
         case MPG123_ENC_FLOAT_32:
-            sh->sample_format = AF_FORMAT_FLOAT_NE;
+            sh->sample_format = AF_FORMAT_FLOAT;
             break;
         default:
             /* This means we got a funny custom build of libmpg123 that only supports an unknown format. */
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index 769a2b4577..c4f826e856 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -135,7 +135,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 break;
             default:
                 af->play = play_f;
-                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
                 break;
         }
 
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index ed482c7a6b..b64d5b54bd 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -50,7 +50,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
     mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
 
     return af_test_output(af,(struct mp_audio*)arg);
   }
diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c
index ee8aff5afc..0c3e6f8497 100644
--- a/audio/filter/af_convert24.c
+++ b/audio/filter/af_convert24.c
@@ -23,10 +23,10 @@
 
 static bool test_conversion(int src_format, int dst_format)
 {
-    return (src_format == AF_FORMAT_U24_NE && dst_format == AF_FORMAT_U32_NE) ||
-           (src_format == AF_FORMAT_S24_NE && dst_format == AF_FORMAT_S32_NE) ||
-           (src_format == AF_FORMAT_U32_NE && dst_format == AF_FORMAT_U24_NE) ||
-           (src_format == AF_FORMAT_S32_NE && dst_format == AF_FORMAT_S24_NE);
+    return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) ||
+           (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
+           (src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) ||
+           (src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
 }
 
 static int control(struct af_instance *af, int cmd, void *arg)
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index 17c4a12a95..685dbcd8d5 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -91,8 +91,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     mp_audio_force_interleaved_format((struct mp_audio*)arg);
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
 
-    if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16_NE)){
-      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16)){
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
     }
     return af_test_output(af,(struct mp_audio*)arg);
   case AF_CONTROL_COMMAND_LINE:{
@@ -296,14 +296,14 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
 {
   af_drc_t *s = af->setup;
 
-  if(af->data->format == (AF_FORMAT_S16_NE))
+  if(af->data->format == (AF_FORMAT_S16))
   {
     if (s->method)
 	method2_int16(s, data);
     else
 	method1_int16(s, data);
   }
-  else if(af->data->format == (AF_FORMAT_FLOAT_NE))
+  else if(af->data->format == (AF_FORMAT_FLOAT))
   {
     if (s->method)
 	method2_float(s, data);
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index 75a489d867..6aff565607 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -98,7 +98,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
 
     // Calculate number of active filters
     s->K=KM;
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index 5e1096f85a..6c1ea6459b 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -88,7 +88,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     // Accept only int16_t as input format (which sucks)
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
 
     // If buffer length isn't set, set it to the default value
     if(s->sz == 0)
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index 4561b60690..6a00fb7e65 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -51,12 +51,12 @@ static int control(struct af_instance* af, int cmd, void* arg)
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
     mp_audio_force_interleaved_format(af->data);
     mp_audio_set_num_channels(af->data, 2);
-    if (af->data->format == AF_FORMAT_FLOAT_NE)
+    if (af->data->format == AF_FORMAT_FLOAT)
     {
 	af->play = play_float;
     }// else
     {
-        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+        mp_audio_set_format(af->data, AF_FORMAT_S16);
 	af->play = play_s16;
     }
 
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index ed51351750..bfb619b040 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -311,7 +311,7 @@ static int control(struct af_instance *af, int cmd, void* arg)
 	    }
 	    else if (af->data->nch < 5)
 	      mp_audio_set_channels_old(af->data, 5);
-        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+        mp_audio_set_format(af->data, AF_FORMAT_S16);
 	test_output_res = af_test_output(af, (struct mp_audio*)arg);
 	// after testing input set the real output format
         mp_audio_set_num_channels(af->data, 2);
diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c
index 8c633b136c..07ef0579bc 100644
--- a/audio/filter/af_karaoke.c
+++ b/audio/filter/af_karaoke.c
@@ -35,7 +35,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 	switch(cmd){
 		case AF_CONTROL_REINIT:
                 mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
 		return af_test_output(af,(struct mp_audio*)arg);
 	}
 	return AF_UNKNOWN;
diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c
index df88c06ab2..50d0bb5d85 100644
--- a/audio/filter/af_ladspa.c
+++ b/audio/filter/af_ladspa.c
@@ -495,7 +495,7 @@ static int control(struct af_instance *af, int cmd, void *arg) {
         /* accept FLOAT, let af_format do conversion */
 
         mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-        mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+        mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
 
         return af_test_output(af, (struct mp_audio*)arg);
     case AF_CONTROL_COMMAND_LINE: {
diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c
index 24ff8c5985..57a1055149 100644
--- a/audio/filter/af_lavfi.c
+++ b/audio/filter/af_lavfi.c
@@ -175,7 +175,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         struct mp_audio *out = af->data;
 
         if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
-            mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
+            mp_audio_set_format(in, AF_FORMAT_FLOAT);
 
         if (!mp_chmap_is_lavc(&in->channels))
             mp_chmap_reorder_to_lavc(&in->channels); // will always work
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index 2e298a3f39..9b4d8ca586 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -245,7 +245,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
             mp_audio_set_channels(out, &in->channels);
 
         if (af_to_avformat(in->format) == AV_SAMPLE_FMT_NONE)
-            mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
+            mp_audio_set_format(in, AF_FORMAT_FLOAT);
         if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE)
             mp_audio_set_format(out, in->format);
 
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index 3d8c6045d0..29d38c3860 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -55,7 +55,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
     set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch);
 
     if((af->data->format != ((struct mp_audio*)arg)->format) ||
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index c8560af502..659b7971f5 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -290,10 +290,10 @@ static int control(struct af_instance *af, int cmd, void *arg)
             return af_test_output(af, data);
         }
 
-        if (data->format == AF_FORMAT_S16_NE) {
+        if (data->format == AF_FORMAT_S16) {
             use_int = 1;
         } else {
-            mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
         }
         int bps = af->data->bps;
 
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index ef6fd7d37b..f241c5475a 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -57,15 +57,15 @@ static int control(struct af_instance* af, int cmd, void* arg)
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
     mp_audio_set_num_channels(af->data, 1);
 #if 0
-    if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
+    if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT)
     {
-	af->data->format = AF_FORMAT_FLOAT_NE;
+	af->data->format = AF_FORMAT_FLOAT;
 	af->data->bps = 4;
 	af->play = play_float;
     }// else
 #endif
     {
-        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+        mp_audio_set_format(af->data, AF_FORMAT_S16);
 	af->play = play_s16;
     }
 
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 4fd16904c9..cdb4c04ea3 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -72,7 +72,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
     mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
-    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
 
     // Design low-pass filter
     s->k = 1.0;
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c
index efeecdc1e3..f06789eabe 100644
--- a/audio/filter/af_surround.c
+++ b/audio/filter/af_surround.c
@@ -98,7 +98,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
         return AF_DETACH;
     }
 
-    mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_format(in, AF_FORMAT_FLOAT);
     mp_audio_copy_config(af->data, in);
     mp_audio_set_channels_old(af->data, in->nch * 2);
 
diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c
index c153d4261a..a06cad7600 100644
--- a/audio/filter/af_sweep.c
+++ b/audio/filter/af_sweep.c
@@ -42,7 +42,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
   switch(cmd){
   case AF_CONTROL_REINIT:
     mp_audio_copy_config(af->data, data);
-    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
 
     return af_test_output(af, data);
   case AF_CONTROL_COMMAND_LINE:
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
index b0eef1865c..edf29d00f2 100644
--- a/audio/filter/af_volume.c
+++ b/audio/filter/af_volume.c
@@ -47,10 +47,10 @@ static int control(struct af_instance *af, int cmd, void *arg)
         mp_audio_copy_config(af->data, in);
         mp_audio_force_interleaved_format(af->data);
 
-        if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT_NE) {
-            mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
+        if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT) {
+            mp_audio_set_format(af->data, AF_FORMAT_S16);
         } else {
-            mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
         }
         if (af_fmt_is_planar(in->format))
             mp_audio_set_format(af->data, af_fmt_to_planar(af->data->format));
@@ -70,7 +70,7 @@ static void filter_plane(struct af_instance *af, void *ptr, int num_samples)
 {
     struct priv *s = af->priv;
 
-    if (af_fmt_from_planar(af->data->format) == AF_FORMAT_S16_NE) {
+    if (af_fmt_from_planar(af->data->format) == AF_FORMAT_S16) {
         int16_t *a = ptr;
         int vol = 256.0 * s->level;
         if (vol != 256) {
@@ -79,7 +79,7 @@ static void filter_plane(struct af_instance *af, void *ptr, int num_samples)
                 a[i] = MPCLAMP(x, SHRT_MIN, SHRT_MAX);
             }
         }
-    } else if (af_fmt_from_planar(af->data->format) == AF_FORMAT_FLOAT_NE) {
+    } else if (af_fmt_from_planar(af->data->format) == AF_FORMAT_FLOAT) {
         float *a = ptr;
         float vol = s->level;
         if (vol != 1.0) {
diff --git a/audio/fmt-conversion.c b/audio/fmt-conversion.c
index 93fda3eaa0..da770a8eda 100644
--- a/audio/fmt-conversion.c
+++ b/audio/fmt-conversion.c
@@ -27,10 +27,10 @@ static const struct {
     int fmt;
 } audio_conversion_map[] = {
     {AV_SAMPLE_FMT_U8,    AF_FORMAT_U8},
-    {AV_SAMPLE_FMT_S16,   AF_FORMAT_S16_NE},
-    {AV_SAMPLE_FMT_S32,   AF_FORMAT_S32_NE},
-    {AV_SAMPLE_FMT_FLT,   AF_FORMAT_FLOAT_NE},
-    {AV_SAMPLE_FMT_DBL,   AF_FORMAT_DOUBLE_NE},
+    {AV_SAMPLE_FMT_S16,   AF_FORMAT_S16},
+    {AV_SAMPLE_FMT_S32,   AF_FORMAT_S32},
+    {AV_SAMPLE_FMT_FLT,   AF_FORMAT_FLOAT},
+    {AV_SAMPLE_FMT_DBL,   AF_FORMAT_DOUBLE},
 
     {AV_SAMPLE_FMT_U8P,   AF_FORMAT_U8P},
     {AV_SAMPLE_FMT_S16P,  AF_FORMAT_S16P},
diff --git a/audio/format.c b/audio/format.c
index d0cd04cb88..3c22de476a 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -109,7 +109,6 @@ bool af_fmt_is_planar(int format)
 
 #define FMT_ENDIAN(string, id)                          \
     {string,            id},                            \
-    {string "ne",       id},                            \
     {string "le",       MP_CONCAT(id, _LE)},            \
     {string "be",       MP_CONCAT(id, _BE)},            \
 
diff --git a/audio/format.h b/audio/format.h
index 9b855e4689..43f0da33f6 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -130,17 +130,6 @@ enum af_format {
     AF_FORMAT_IEC61937  = AF_SELECT_LE_BE(AF_FORMAT_IEC61937_LE, AF_FORMAT_IEC61937_BE),
 };
 
-#define AF_FORMAT_U16_NE AF_FORMAT_U16
-#define AF_FORMAT_S16_NE AF_FORMAT_S16
-#define AF_FORMAT_U24_NE AF_FORMAT_U24
-#define AF_FORMAT_S24_NE AF_FORMAT_S24
-#define AF_FORMAT_U32_NE AF_FORMAT_U32
-#define AF_FORMAT_S32_NE AF_FORMAT_S32
-#define AF_FORMAT_FLOAT_NE AF_FORMAT_FLOAT
-#define AF_FORMAT_DOUBLE_NE AF_FORMAT_DOUBLE
-#define AF_FORMAT_AC3_NE AF_FORMAT_AC3
-#define AF_FORMAT_IEC61937_NE AF_FORMAT_IEC61937
-
 #define AF_FORMAT_IS_AC3(fmt) \
     (((fmt) & AF_FORMAT_SPECIAL_MASK) == AF_FORMAT_S_AC3)
 
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 86c813c143..a8df1c9f25 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -387,7 +387,7 @@ static int init(struct ao *ao)
                    ao->channels.num);
     } else {
         device = select_chmap(ao);
-        if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT_NE)
+        if (strcmp(device, "default") != 0 && ao->format == AF_FORMAT_FLOAT)
         {
             // hack - use the converter plugin (why the heck?)
             device = talloc_asprintf(ao, "plug:%s", device);
@@ -435,7 +435,7 @@ static int init(struct ao *ao)
     p->alsa_fmt = find_alsa_format(ao->format);
     if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
         p->alsa_fmt = SND_PCM_FORMAT_S16;
-        ao->format = AF_FORMAT_S16_NE;
+        ao->format = AF_FORMAT_S16;
     }
 
     err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index f828a210dc..ec5e83bd50 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -392,7 +392,7 @@ static int init(struct ao *ao)
     int rate = ao->samplerate;
 
     if (AF_FORMAT_IS_AC3(format))
-        format = AF_FORMAT_AC3_NE;
+        format = AF_FORMAT_AC3;
     else {
         struct mp_chmap_sel sel = {0};
         mp_chmap_sel_add_waveext(&sel);
@@ -400,7 +400,7 @@ static int init(struct ao *ao)
             return -1;
     }
     switch (format) {
-    case AF_FORMAT_AC3_NE:
+    case AF_FORMAT_AC3:
     case AF_FORMAT_S24_LE:
     case AF_FORMAT_S16_LE:
     case AF_FORMAT_U8:
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index 09a2951629..b1f2028af2 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -89,14 +89,14 @@ static int format_table[][2] = {
     {AFMT_S32_BE,       AF_FORMAT_S32_BE},
 #endif
 #ifdef AFMT_FLOAT
-    {AFMT_FLOAT,        AF_FORMAT_FLOAT_NE},
+    {AFMT_FLOAT,        AF_FORMAT_FLOAT},
 #endif
     // SPECIALS
 #ifdef AFMT_MPEG
     {AFMT_MPEG,         AF_FORMAT_MPEG2},
 #endif
 #ifdef AFMT_AC3
-    {AFMT_AC3,          AF_FORMAT_AC3_NE},
+    {AFMT_AC3,          AF_FORMAT_AC3},
 #endif
     {-1, -1}
 };
@@ -269,7 +269,7 @@ static int init(struct ao *ao)
 
 ac3_retry:
     if (AF_FORMAT_IS_AC3(ao->format))
-        ao->format = AF_FORMAT_AC3_NE;
+        ao->format = AF_FORMAT_AC3;
     oss_format = format2oss(ao->format);
     if (oss_format == -1) {
         MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
@@ -279,15 +279,15 @@ ac3_retry:
 #else
         oss_format = AFMT_S16_LE;
 #endif
-        ao->format = AF_FORMAT_S16_NE;
+        ao->format = AF_FORMAT_S16;
     }
     if (ioctl(p->audio_fd, SNDCTL_DSP_SETFMT, &oss_format) < 0 ||
         oss_format != format2oss(ao->format))
     {
         MP_WARN(ao, "Can't set audio device %s to %s output, trying %s...\n",
                 p->dsp, af_fmt_to_str(ao->format),
-                af_fmt_to_str(AF_FORMAT_S16_NE));
-        ao->format = AF_FORMAT_S16_NE;
+                af_fmt_to_str(AF_FORMAT_S16));
+        ao->format = AF_FORMAT_S16;
         goto ac3_retry;
     }
 
diff --git a/audio/out/ao_portaudio.c b/audio/out/ao_portaudio.c
index 9c0d7804f8..fad1dc12d8 100644
--- a/audio/out/ao_portaudio.c
+++ b/audio/out/ao_portaudio.c
@@ -56,12 +56,12 @@ struct format_map {
 
 static const struct format_map format_maps[] = {
     // first entry is the default format
-    {AF_FORMAT_S16_NE,      paInt16},
-    {AF_FORMAT_S24_NE,      paInt24},
-    {AF_FORMAT_S32_NE,      paInt32},
-    {AF_FORMAT_S8,          paInt8},
-    {AF_FORMAT_U8,          paUInt8},
-    {AF_FORMAT_FLOAT_NE,    paFloat32},
+    {AF_FORMAT_S16,      paInt16},
+    {AF_FORMAT_S24,      paInt24},
+    {AF_FORMAT_S32,      paInt32},
+    {AF_FORMAT_S8,       paInt8},
+    {AF_FORMAT_U8,       paUInt8},
+    {AF_FORMAT_FLOAT,    paFloat32},
     {AF_FORMAT_UNKNOWN,     0}
 };