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| author | Kővágó, Zoltán <dirty.ice.hu@gmail.com> | 2019-03-08 23:34:13 +0100 |
|---|---|---|
| committer | Gerd Hoffmann <kraxel@redhat.com> | 2019-03-11 10:29:26 +0100 |
| commit | 85bc58520c0e43660cbbe51b9eb5022a0baafe9f (patch) | |
| tree | 6a6e20f651bcb5ae047e90ed823d2dcaa10e06e1 /hw/audio/cs4231a.c | |
| parent | 8c3a7d008794305b1304549f1d9249c12cbf5b2b (diff) | |
| download | qemu-85bc58520c0e43660cbbe51b9eb5022a0baafe9f.zip qemu-85bc58520c0e43660cbbe51b9eb5022a0baafe9f.tar.gz qemu-85bc58520c0e43660cbbe51b9eb5022a0baafe9f.tar.bz2 | |
audio: use qapi AudioFormat instead of audfmt_e
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>
Diffstat (limited to 'hw/audio/cs4231a.c')
| -rw-r--r-- | hw/audio/cs4231a.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c index 9089dcb..62da75e 100644 --- a/hw/audio/cs4231a.c +++ b/hw/audio/cs4231a.c @@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) { case 0: - as.fmt = AUD_FMT_U8; + as.fmt = AUDIO_FORMAT_U8; s->shift = as.nchannels == 2; break; @@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) case 3: s->tab = ALawDecompressTable; x_law: - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = AUDIO_HOST_ENDIANNESS; s->shift = as.nchannels == 2; break; @@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val) as.endianness = 1; /* fall through */ case 2: - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; s->shift = as.nchannels; break; |