audio: replace af_fmt2str_short -> af_fmt_to_str

Also, remove all af_fmt2str usages.

13 files changed, 22 insertions, 34 deletions

diff --git a/audio/audio.c b/audio/audio.c
index e1d3a76978..9d41928436 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -65,7 +65,7 @@ char *mp_audio_fmt_to_str(int srate, const struct mp_chmap *chmap, int format)
 {
     char *chstr = mp_chmap_to_str(chmap);
     char *res = talloc_asprintf(NULL, "%dHz %s %dch %s", srate, chstr,
-                                chmap->num, af_fmt2str_short(format));
+                                chmap->num, af_fmt_to_str(format));
     talloc_free(chstr);
     return res;
 }
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index b9ca71692f..127139ff60 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -160,7 +160,7 @@ int init_best_audio_codec(sh_audio_t *sh_audio, char *audio_decoders)
         mp_msg(MSGT_DECAUDIO, MSGL_V,
                "AUDIO: %d Hz, %d ch, %s\n",
                sh_audio->samplerate, sh_audio->channels.num,
-               af_fmt2str_short(sh_audio->sample_format));
+               af_fmt_to_str(sh_audio->sample_format));
         mp_msg(MSGT_IDENTIFY, MSGL_INFO,
                "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
                sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels.num);
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index aa92b8be23..0e77b3e4eb 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -74,7 +74,6 @@ static int control(struct af_instance *af, int cmd, void *arg)
     switch (cmd) {
     case AF_CONTROL_REINIT: {
         int format;
-        char buf[256];
         // Sanity check
         if (!arg) return AF_ERROR;
 
@@ -150,7 +149,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         }
         bs2b_set_srate(s->filter, (long)af->data->rate);
         mp_msg(MSGT_AFILTER, MSGL_V, "[bs2b] using format %s\n",
-               af_fmt2str(af->data->format,buf,256));
+               af_fmt_to_str(af->data->format));
 
         return af_test_output(af,(struct mp_audio*)arg);
     }
diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c
index 03c9aa9e99..ab601ba9bb 100644
--- a/audio/filter/af_dummy.c
+++ b/audio/filter/af_dummy.c
@@ -34,7 +34,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
   case AF_CONTROL_REINIT: ;
     *af->data = *(struct mp_audio*)arg;
     mp_msg(MSGT_AFILTER, MSGL_V, "[dummy] Was reinitialized: %iHz/%ich/%s\n",
-	af->data->rate,af->data->nch,af_fmt2str_short(af->data->format));
+	af->data->rate,af->data->nch,af_fmt_to_str(af->data->format));
     return AF_OK;
   }
   return AF_UNKNOWN;
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index cd3548ea8f..9ca193017a 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -362,9 +362,8 @@ static int af_open(struct af_instance* af){
             break;
         }
     }
-    char buf[100];
     mp_msg(MSGT_AFILTER, MSGL_V, "[af_lavcac3enc]: in sample format: %s\n",
-           af_fmt2str(s->in_sampleformat, buf, 100));
+           af_fmt_to_str(s->in_sampleformat));
     s->pending_data_size = AF_NCH * AC3_FRAME_SIZE *
         af_fmt2bits(s->in_sampleformat) / 8;
     s->pending_data = malloc(s->pending_data_size);
diff --git a/audio/fmt-conversion.c b/audio/fmt-conversion.c
index ce2a9054bf..bb32361ce5 100644
--- a/audio/fmt-conversion.c
+++ b/audio/fmt-conversion.c
@@ -45,7 +45,7 @@ enum AVSampleFormat af_to_avformat(int fmt)
     sample_fmt = audio_conversion_map[i].sample_fmt;
     if (sample_fmt == AF_FORMAT_UNKNOWN)
         mp_msg(MSGT_GLOBAL, MSGL_V, "Unsupported sample format: %s\n",
-                af_fmt2str_short(fmt));
+                af_fmt_to_str(fmt));
     return sample_fmt;
 }
 
diff --git a/audio/format.c b/audio/format.c
index 9b0967ae53..01c9a431a1 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -62,15 +62,6 @@ int af_fmt_change_bits(int format, int bits)
     return af_fmt_is_valid(format) ? format : 0;
 }
 
-/* Convert format to str input str is a buffer for the
-   converted string, size is the size of the buffer */
-char *af_fmt2str(int format, char* str, int size)
-{
-    const char *name = af_fmt2str_short(format);
-    snprintf(str, size, "%s", name);
-    return str;
-}
-
 const struct af_fmt_entry af_fmtstr_table[] = {
     { "mpeg2", AF_FORMAT_MPEG2 },
     { "ac3le", AF_FORMAT_AC3_LE },
@@ -119,7 +110,7 @@ bool af_fmt_is_valid(int format)
     return false;
 }
 
-const char *af_fmt2str_short(int format)
+const char *af_fmt_to_str(int format)
 {
     for (int i = 0; af_fmtstr_table[i].name; i++) {
         if (af_fmtstr_table[i].format == format)
diff --git a/audio/format.h b/audio/format.h
index 9a898fe55c..6ec435b6b7 100644
--- a/audio/format.h
+++ b/audio/format.h
@@ -130,15 +130,14 @@ struct af_fmt_entry {
 extern const struct af_fmt_entry af_fmtstr_table[];
 
 int af_str2fmt_short(bstr str);
+const char *af_fmt_to_str(int format);
+
 int af_fmt2bits(int format);
 int af_fmt_change_bits(int format, int bits);
 
 // Amount of bytes that contain audio of the given duration, aligned to frames.
 int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate);
 
-char *af_fmt2str(int format, char* str, int size);
-const char *af_fmt2str_short(int format);
-
 bool af_fmt_is_valid(int format);
 
 #endif /* MPLAYER_AF_FORMAT_H */
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,