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author | Kővágó, Zoltán <dirty.ice.hu@gmail.com> | 2019-03-08 23:34:13 +0100 |
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committer | Gerd Hoffmann <kraxel@redhat.com> | 2019-03-11 10:29:26 +0100 |
commit | 85bc58520c0e43660cbbe51b9eb5022a0baafe9f (patch) | |
tree | 6a6e20f651bcb5ae047e90ed823d2dcaa10e06e1 /hw/audio | |
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')
-rw-r--r-- | hw/audio/ac97.c | 2 | ||||
-rw-r--r-- | hw/audio/adlib.c | 2 | ||||
-rw-r--r-- | hw/audio/cs4231a.c | 6 | ||||
-rw-r--r-- | hw/audio/es1370.c | 4 | ||||
-rw-r--r-- | hw/audio/gus.c | 2 | ||||
-rw-r--r-- | hw/audio/hda-codec.c | 18 | ||||
-rw-r--r-- | hw/audio/lm4549.c | 6 | ||||
-rw-r--r-- | hw/audio/milkymist-ac97.c | 2 | ||||
-rw-r--r-- | hw/audio/pcspk.c | 2 | ||||
-rw-r--r-- | hw/audio/sb16.c | 14 | ||||
-rw-r--r-- | hw/audio/wm8750.c | 6 |
11 files changed, 32 insertions, 32 deletions
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c index d799533..2265622 100644 --- a/hw/audio/ac97.c +++ b/hw/audio/ac97.c @@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq) as.freq = freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; if (freq > 0) { diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c index 97b876c..0957780 100644 --- a/hw/audio/adlib.c +++ b/hw/audio/adlib.c @@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = SHIFT; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = AUDIO_HOST_ENDIANNESS; AUD_register_card ("adlib", &s->card); 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; diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c index 97789a0..a5314d6 100644 --- a/hw/audio/es1370.c +++ b/hw/audio/es1370.c @@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl) i, new_freq, 1 << (new_fmt & 1), - (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8, + (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8, d->shift); if (new_freq) { struct audsettings as; as.freq = new_freq; as.nchannels = 1 << (new_fmt & 1); - as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8; + as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; as.endianness = 0; if (i == ADC_CHANNEL) { diff --git a/hw/audio/gus.c b/hw/audio/gus.c index 8e0b27e..b3e2a7f 100644 --- a/hw/audio/gus.c +++ b/hw/audio/gus.c @@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp) as.freq = s->freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = GUS_ENDIANNESS; s->voice = AUD_open_out ( diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c index 617a1c1..c25bfa3 100644 --- a/hw/audio/hda-codec.c +++ b/hw/audio/hda-codec.c @@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as) } switch (format & AC_FMT_BITS_MASK) { - case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break; - case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break; - case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break; + case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break; + case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break; + case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break; } as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1; @@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as) /* -------------------------------------------------------------------------- */ static const char *fmt2name[] = { - [ AUD_FMT_U8 ] = "PCM-U8", - [ AUD_FMT_S8 ] = "PCM-S8", - [ AUD_FMT_U16 ] = "PCM-U16", - [ AUD_FMT_S16 ] = "PCM-S16", - [ AUD_FMT_U32 ] = "PCM-U32", - [ AUD_FMT_S32 ] = "PCM-S32", + [ AUDIO_FORMAT_U8 ] = "PCM-U8", + [ AUDIO_FORMAT_S8 ] = "PCM-S8", + [ AUDIO_FORMAT_U16 ] = "PCM-U16", + [ AUDIO_FORMAT_S16 ] = "PCM-S16", + [ AUDIO_FORMAT_U32 ] = "PCM-U32", + [ AUDIO_FORMAT_S32 ] = "PCM-S32", }; typedef struct HDAAudioState HDAAudioState; diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c index a46f230..af8b22b 100644 --- a/hw/audio/lm4549.c +++ b/hw/audio/lm4549.c @@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s, struct audsettings as; as.freq = value; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( @@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id) struct audsettings as; as.freq = freq; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( @@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque) /* Open a default voice */ as.freq = 48000; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; s->voice = AUD_open_out( diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c index bc8db71..90cce1e 100644 --- a/hw/audio/milkymist-ac97.c +++ b/hw/audio/milkymist-ac97.c @@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp) as.freq = 48000; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 1; s->voice_in = AUD_open_in(&s->card, s->voice_in, diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c index b80a62c..fdbb4b6 100644 --- a/hw/audio/pcspk.c +++ b/hw/audio/pcspk.c @@ -162,7 +162,7 @@ static void pcspk_initfn(Object *obj) static void pcspk_realizefn(DeviceState *dev, Error **errp) { - struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0}; + struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0}; ISADevice *isadev = ISA_DEVICE(dev); PCSpkState *s = PC_SPEAKER(dev); diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c index c5b9bf7..65ea0cd 100644 --- a/hw/audio/sb16.c +++ b/hw/audio/sb16.c @@ -66,7 +66,7 @@ typedef struct SB16State { int fmt_stereo; int fmt_signed; int fmt_bits; - audfmt_e fmt; + AudioFormat fmt; int dma_auto; int block_size; int fifo; @@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s) static void dma_cmd8 (SB16State *s, int mask, int dma_len) { - s->fmt = AUD_FMT_U8; + s->fmt = AUDIO_FORMAT_U8; s->use_hdma = 0; s->fmt_bits = 8; s->fmt_signed = 0; @@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len) if (16 == s->fmt_bits) { if (s->fmt_signed) { - s->fmt = AUD_FMT_S16; + s->fmt = AUDIO_FORMAT_S16; } else { - s->fmt = AUD_FMT_U16; + s->fmt = AUDIO_FORMAT_U16; } } else { if (s->fmt_signed) { - s->fmt = AUD_FMT_S8; + s->fmt = AUDIO_FORMAT_S8; } else { - s->fmt = AUD_FMT_U8; + s->fmt = AUDIO_FORMAT_U8; } } @@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s) as.freq = s->freq; as.nchannels = 1; - as.fmt = AUD_FMT_U8; + as.fmt = AUDIO_FORMAT_U8; as.endianness = 0; s->voice = AUD_open_out ( diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c index 169b006..ca0ad73 100644 --- a/hw/audio/wm8750.c +++ b/hw/audio/wm8750.c @@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s) in_fmt.endianness = 0; in_fmt.nchannels = 2; in_fmt.freq = s->adc_hz; - in_fmt.fmt = AUD_FMT_S16; + in_fmt.fmt = AUDIO_FORMAT_S16; s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0], CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt); @@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s) out_fmt.endianness = 0; out_fmt.nchannels = 2; out_fmt.freq = s->dac_hz; - out_fmt.fmt = AUD_FMT_S16; + out_fmt.fmt = AUDIO_FORMAT_S16; s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0], CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt); @@ -681,7 +681,7 @@ uint32_t wm8750_adc_dat(void *opaque) if (s->idx_in >= sizeof(s->data_in)) { wm8750_in_load(s); if (s->idx_in >= sizeof(s->data_in)) { - return 0x80008000; /* silence in AUD_FMT_S16 sample format */ + return 0x80008000; /* silence in AUDIO_FORMAT_S16 sample format */ } } |