5 files changed, 13 insertions, 13 deletions

diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 1ad7598075..1f7dc71d3d 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -442,7 +442,7 @@ static int init(struct ao *ao)
     err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
     if (err < 0) {
         MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
-                af_fmt2str_short(ao->format));
+                af_fmt_to_str(ao->format));
         p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
         if (AF_FORMAT_IS_AC3(ao->format))
             ao->format = AF_FORMAT_AC3_LE;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 034f5d8446..464947c0dc 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -407,7 +407,7 @@ static int init(struct ao *ao)
         break;
     default:
         MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
-                   af_fmt2str_short(format));
+                   af_fmt_to_str(format));
         format = AF_FORMAT_S16_LE;
     }
     //set our audio parameters
@@ -416,7 +416,7 @@ static int init(struct ao *ao)
     ao->bps = ao->channels.num * rate * (af_fmt2bits(format) >> 3);
     int buffersize = ao->bps; // space for 1 sec
     MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
-               ao->channels.num, af_fmt2str_short(format));
+               ao->channels.num, af_fmt_to_str(format));
     MP_VERBOSE(ao, "Buffersize:%d bytes (%d msec)\n",
                buffersize, buffersize / ao->bps * 1000);
 
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index 283b960255..1a28267613 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -205,7 +205,7 @@ static int init(struct ao *ao)
         mchan = p->cfg_oss_mixer_channel;
 
     MP_VERBOSE(ao, "%d Hz  %d chans  %s\n", ao->samplerate,
-               ao->channels.num, af_fmt2str_short(ao->format));
+               ao->channels.num, af_fmt_to_str(ao->format));
 
     if (mchan) {
         int fd, devs, i;
@@ -274,7 +274,7 @@ ac3_retry:
     oss_format = format2oss(ao->format);
     if (oss_format == -1) {
         MP_VERBOSE(ao, "Unknown/not supported internal format: %s\n",
-                   af_fmt2str_short(ao->format));
+                   af_fmt_to_str(ao->format));
 #if BYTE_ORDER == BIG_ENDIAN
         oss_format = AFMT_S16_BE;
 #else
@@ -286,8 +286,8 @@ ac3_retry:
         oss_format != format2oss(ao->format))
     {
         MP_WARN(ao, "Can't set audio device %s to %s output, trying %s...\n",
-                p->dsp, af_fmt2str_short(ao->format),
-                af_fmt2str_short(AF_FORMAT_S16_NE));
+                p->dsp, af_fmt_to_str(ao->format),
+                af_fmt_to_str(AF_FORMAT_S16_NE));
         ao->format = AF_FORMAT_S16_NE;
         goto ac3_retry;
     }
@@ -298,7 +298,7 @@ ac3_retry:
         return -1;
     }
 
-    MP_VERBOSE(ao, "sample format: %s\n", af_fmt2str_short(ao->format));
+    MP_VERBOSE(ao, "sample format: %s\n", af_fmt_to_str(ao->format));
 
     if (!AF_FORMAT_IS_AC3(ao->format)) {
         struct mp_chmap_sel sel = {0};
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index d2f5f791ec..f7d793700d 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -145,7 +145,7 @@ static int init(struct ao *ao)
     MP_INFO(ao, "File: %s (%s)\nPCM: Samplerate: %d Hz Channels: %d Format: %s\n",
             priv->outputfilename,
             priv->waveheader ? "WAVE" : "RAW PCM", ao->samplerate,
-            ao->channels.num, af_fmt2str_short(ao->format));
+            ao->channels.num, af_fmt_to_str(ao->format));
     MP_INFO(ao, "Info: Faster dumping is achieved with -no-video\n");
     MP_INFO(ao, "Info: To write WAVE files use -ao pcm:waveheader (default).\n");
 
diff --git a/audio/out/ao_wasapi.c b/audio/out/ao_wasapi.c
index 47c6fcfdb7..c605e1cd5d 100644
--- a/audio/out/ao_wasapi.c
+++ b/audio/out/ao_wasapi.c
@@ -325,7 +325,7 @@ static int try_format(struct wasapi_state *state,
 
     EnterCriticalSection(&state->print_lock);
     mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: trying %dch %s @ %dhz\n",
-           channels.num, af_fmt2str_short(af_format), samplerate);
+           channels.num, af_fmt_to_str(af_format), samplerate);
     LeaveCriticalSection(&state->print_lock);
 
     union WAVEFMT u;
@@ -351,7 +351,7 @@ static int try_format(struct wasapi_state *state,
         if (set_ao_format(state, ao, wformat)) {
             EnterCriticalSection(&state->print_lock);
             mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: accepted as %dch %s @ %dhz\n",
-                ao->channels.num, af_fmt2str_short(ao->format), ao->samplerate);
+                ao->channels.num, af_fmt_to_str(ao->format), ao->samplerate);
             LeaveCriticalSection(&state->print_lock);
 
             return 1;
@@ -361,7 +361,7 @@ static int try_format(struct wasapi_state *state,
         if (set_ao_format(state, ao, wformat)) {
             EnterCriticalSection(&state->print_lock);
             mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: %dch %s @ %dhz accepted\n",
-                   ao->channels.num, af_fmt2str_short(af_format), samplerate);
+                   ao->channels.num, af_fmt_to_str(af_format), samplerate);
             LeaveCriticalSection(&state->print_lock);
             return 1;
         }
@@ -418,7 +418,7 @@ static int try_passthrough(struct wasapi_state *state,
 
     EnterCriticalSection(&state->print_lock);
     mp_msg(MSGT_AO, MSGL_V, "ao-wasapi: trying passthrough for %s...\n",
-        af_fmt2str_short((ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE));
+           af_fmt_to_str((ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE));
     LeaveCriticalSection(&state->print_lock);
 
     HRESULT hr = IAudioClient_IsFormatSupported(state->pAudioClient,