audio: fix format function consistency issues

Replace all the check macros with function calls. Give them all the
same case and naming schema.

Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().

Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".

1 files changed, 5 insertions, 5 deletions

diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 8b1e10a10b..c581bf512e 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -444,7 +444,7 @@ static int init(struct ao *ao)
     int format = af_fmt_from_planar(ao->format);
     int rate = ao->samplerate;
 
-    if (!AF_FORMAT_IS_IEC61937(format)) {
+    if (!af_fmt_is_spdif(format)) {
         struct mp_chmap_sel sel = {0};
         mp_chmap_sel_add_waveext(&sel);
         if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
@@ -456,7 +456,7 @@ static int init(struct ao *ao)
     case AF_FORMAT_U8:
         break;
     default:
-        if (AF_FORMAT_IS_IEC61937(format))
+        if (af_fmt_is_spdif(format))
             break;
         MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
                    af_fmt_to_str(format));
@@ -465,7 +465,7 @@ static int init(struct ao *ao)
     //set our audio parameters
     ao->samplerate = rate;
     ao->format = format;
-    ao->bps = ao->channels.num * rate * af_fmt2bps(format);
+    ao->bps = ao->channels.num * rate * af_fmt_to_bytes(format);
     int buffersize = ao->bps * p->cfg_buffersize / 1000;
     MP_VERBOSE(ao, "Samplerate:%iHz Channels:%i Format:%s\n", rate,
                ao->channels.num, af_fmt_to_str(format));
@@ -478,7 +478,7 @@ static int init(struct ao *ao)
                     ? sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX) : 0;
     wformat.Format.nChannels = ao->channels.num;
     wformat.Format.nSamplesPerSec = rate;
-    if (AF_FORMAT_IS_IEC61937(format)) {
+    if (af_fmt_is_spdif(format)) {
         // Whether it also works with e.g. DTS is unknown, but probably does.
         wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
         wformat.Format.wBitsPerSample = 16;
@@ -486,7 +486,7 @@ static int init(struct ao *ao)
     } else {
         wformat.Format.wFormatTag = (ao->channels.num > 2)
                                     ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-        int bps = af_fmt2bps(format);
+        int bps = af_fmt_to_bytes(format);
         wformat.Format.wBitsPerSample = bps * 8;
         wformat.Format.nBlockAlign = wformat.Format.nChannels * bps;
     }