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audio: use qapi AudioFormat instead of audfmt_e

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I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum.  This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.

This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Kővágó, Zoltán 2019-03-08 23:34:13 +01:00 • committed by Gerd Hoffmann
parent 8c3a7d0087
commit 85bc58520c
26 changed files with 196 additions and 189 deletions
Download patch file Download diff file Expand all files Collapse all files

53
audio/alsaaudio.c
View file

@ -87,7 +87,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
audfmt_e fmt;
AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
case AUD_FMT_S8:
case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
case AUD_FMT_U8:
case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
case AUD_FMT_S16:
case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S16_LE;
}
case AUD_FMT_U16:
case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_U16_LE;
}
case AUD_FMT_S32:
case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S32_LE;
}
case AUD_FMT_U32:
case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
}
}
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
*fmt = AUD_FMT_S8;
*fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
*fmt = AUD_FMT_U8;
*fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
*fmt = AUD_FMT_S16;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
*fmt = AUD_FMT_U16;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
*fmt = AUD_FMT_S16;
*fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
*fmt = AUD_FMT_U16;
*fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
*fmt = AUD_FMT_S32;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
*fmt = AUD_FMT_U32;
*fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
*fmt = AUD_FMT_S32;
*fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
*fmt = AUD_FMT_U32;
*fmt = AUDIO_FORMAT_U32;
break;
default:
@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
case AUDIO_FORMAT_S8:
case AUDIO_FORMAT_U8:
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
case AUDIO_FORMAT_S16:
case AUDIO_FORMAT_U16:
bytes_per_sec <<= 1;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_U32:
bytes_per_sec <<= 2;
break;
default:
abort();
}
threshold = (conf->threshold * bytes_per_sec) / 1000;