audio: drop swapped-endian audio formats

Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.

From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.

This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.

4 files changed, 6 insertions, 6 deletions

diff --git a/stream/ai_alsa1x.c b/stream/ai_alsa1x.c
index bf36443dfe..c279505221 100644
--- a/stream/ai_alsa1x.c
+++ b/stream/ai_alsa1x.c
@@ -51,7 +51,7 @@ int ai_alsa_setup(audio_in_t *ai)
         return -1;
     }
 
-    err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE);
+    err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16);
     if (err < 0) {
         MP_ERR(ai, "Sample format not available.\n");
         return -1;
@@ -122,7 +122,7 @@ int ai_alsa_setup(audio_in_t *ai)
         snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
     }
 
-    ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
+    ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16);
     ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
     ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
     ai->samplesize = ai->alsa.bits_per_sample;
diff --git a/stream/ai_oss.c b/stream/ai_oss.c
index b7a7988bde..98477d10f3 100644
--- a/stream/ai_oss.c
+++ b/stream/ai_oss.c
@@ -100,10 +100,10 @@ int ai_oss_init(audio_in_t *ai)
            ioctl(ai->oss.audio_fd, SNDCTL_DSP_GETFMTS, &ioctl_param));
 
     MP_VERBOSE(ai, "Supported formats: %x\n", ioctl_param);
-    if (!(ioctl_param & AFMT_S16_LE))
+    if (!(ioctl_param & AFMT_S16_NE))
         MP_ERR(ai, "unsupported format\n");
 
-    ioctl_param = AFMT_S16_LE;
+    ioctl_param = AFMT_S16_NE;
     MP_VERBOSE(ai, "ioctl dsp setfmt: %d\n",
            err = ioctl(ai->oss.audio_fd, SNDCTL_DSP_SETFMT, &ioctl_param));
     if (err < 0) {
diff --git a/stream/ai_sndio.c b/stream/ai_sndio.c
index dc3c66279d..2bb47955a4 100644
--- a/stream/ai_sndio.c
+++ b/stream/ai_sndio.c
@@ -15,7 +15,7 @@ int ai_sndio_setup(audio_in_t *ai)
 
     par.bits = 16;
     par.sig = 1;
-    par.le = 1;
+    par.le = SIO_LE_NATIVE;
     par.rchan = ai->req_channels;
     par.rate = ai->req_samplerate;
     par.appbufsz = ai->req_samplerate;  /* 1 sec */
diff --git a/stream/tvi_v4l2.c b/stream/tvi_v4l2.c
index 13067b8cab..f2ec84d4cf 100644
--- a/stream/tvi_v4l2.c
+++ b/stream/tvi_v4l2.c
@@ -922,7 +922,7 @@ static int do_control(priv_t *priv, int cmd, void *arg)
     case TVI_CONTROL_AUD_GET_FORMAT:
         init_audio(priv);
         if (!priv->audio_initialized) return TVI_CONTROL_FALSE;
-        *(int *)arg = AF_FORMAT_S16_LE;
+        *(int *)arg = AF_FORMAT_S16;
         MP_VERBOSE(priv, "%s: get audio format: %d\n",
                info.short_name, *(int *)arg);
         return TVI_CONTROL_TRUE;