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audio: drop swapped-endian audio formats

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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.
This commit is contained in:
wm4
2014-09-23 21:04:37 +02:00
parent 5b5a3d0c46
commit b745c2d005
25 changed files with 256 additions and 368 deletions
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4
stream/ai_alsa1x.c
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@@ -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;