diff options
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 | |
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>
-rw-r--r-- | audio/alsaaudio.c | 53 | ||||
-rw-r--r-- | audio/audio.c | 97 | ||||
-rw-r--r-- | audio/audio.h | 12 | ||||
-rw-r--r-- | audio/audio_win_int.c | 18 | ||||
-rw-r--r-- | audio/ossaudio.c | 30 | ||||
-rw-r--r-- | audio/paaudio.c | 28 | ||||
-rw-r--r-- | audio/sdlaudio.c | 26 | ||||
-rw-r--r-- | audio/spiceaudio.c | 4 | ||||
-rw-r--r-- | audio/wavaudio.c | 17 | ||||
-rw-r--r-- | audio/wavcapture.c | 2 | ||||
-rw-r--r-- | hw/arm/omap2.c | 2 | ||||
-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 | ||||
-rw-r--r-- | hw/display/xlnx_dp.c | 2 | ||||
-rw-r--r-- | hw/input/tsc210x.c | 2 | ||||
-rw-r--r-- | hw/usb/dev-audio.c | 2 | ||||
-rw-r--r-- | ui/vnc.c | 26 |
26 files changed, 196 insertions, 189 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 635be73..5bd0342 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -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; diff --git a/audio/audio.c b/audio/audio.c index 909c817..77216e5 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -113,7 +113,7 @@ static struct { .settings = { .freq = 44100, .nchannels = 2, - .fmt = AUD_FMT_S16, + .fmt = AUDIO_FORMAT_S16, .endianness = AUDIO_HOST_ENDIANNESS, } }, @@ -125,7 +125,7 @@ static struct { .settings = { .freq = 44100, .nchannels = 2, - .fmt = AUD_FMT_S16, + .fmt = AUDIO_FORMAT_S16, .endianness = AUDIO_HOST_ENDIANNESS, } }, @@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s) return r; } -static const char *audio_audfmt_to_string (audfmt_e fmt) +static const char *audio_audfmt_to_string (AudioFormat fmt) { switch (fmt) { - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return "U8"; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: return "U16"; - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return "S8"; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: return "S16"; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: return "U32"; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: return "S32"; + + default: + abort(); } dolog ("Bogus audfmt %d returning S16\n", fmt); return "S16"; } -static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, +static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval, int *defaultp) { if (!strcasecmp (s, "u8")) { *defaultp = 0; - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; } else if (!strcasecmp (s, "u16")) { *defaultp = 0; - return AUD_FMT_U16; + return AUDIO_FORMAT_U16; } else if (!strcasecmp (s, "u32")) { *defaultp = 0; - return AUD_FMT_U32; + return AUDIO_FORMAT_U32; } else if (!strcasecmp (s, "s8")) { *defaultp = 0; - return AUD_FMT_S8; + return AUDIO_FORMAT_S8; } else if (!strcasecmp (s, "s16")) { *defaultp = 0; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; } else if (!strcasecmp (s, "s32")) { *defaultp = 0; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; } else { dolog ("Bogus audio format `%s' using %s\n", @@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, } } -static audfmt_e audio_get_conf_fmt (const char *envname, - audfmt_e defval, +static AudioFormat audio_get_conf_fmt (const char *envname, + AudioFormat defval, int *defaultp) { const char *var = getenv (envname); @@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix, case AUD_OPT_FMT: { - audfmt_e *fmtp = opt->valp; + AudioFormat *fmtp = opt->valp; printf ( "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n", state, @@ -492,7 +495,7 @@ static void audio_process_options (const char *prefix, case AUD_OPT_FMT: { - audfmt_e *fmtp = opt->valp; + AudioFormat *fmtp = opt->valp; *fmtp = audio_get_conf_fmt (optname, *fmtp, &def); } break; @@ -524,22 +527,22 @@ static void audio_print_settings (struct audsettings *as) dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels); switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: AUD_log (NULL, "S8"); break; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: AUD_log (NULL, "U8"); break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: AUD_log (NULL, "S16"); break; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: AUD_log (NULL, "U16"); break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: AUD_log (NULL, "S32"); break; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: AUD_log (NULL, "U32"); break; default: @@ -570,12 +573,12 @@ static int audio_validate_settings (struct audsettings *as) invalid |= as->endianness != 0 && as->endianness != 1; switch (as->fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: - case AUD_FMT_S16: - case AUD_FMT_U16: - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: break; default: invalid = 1; @@ -591,25 +594,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a int bits = 8, sign = 0; switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: sign = 1; /* fall through */ - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: sign = 1; /* fall through */ - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: bits = 16; break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: sign = 1; /* fall through */ - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: bits = 32; break; + + default: + abort(); } return info->freq == as->freq && info->nchannels == as->nchannels @@ -623,24 +629,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) int bits = 8, sign = 0, shift = 0; switch (as->fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: sign = 1; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: break; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: sign = 1; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: bits = 16; shift = 1; break; - case AUD_FMT_S32: + case AUDIO_FORMAT_S32: sign = 1; - case AUD_FMT_U32: + case AUDIO_FORMAT_U32: bits = 32; shift = 2; break; + + default: + abort(); } info->freq = as->freq; diff --git a/audio/audio.h b/audio/audio.h index f4339a1..02f29a3 100644 --- a/audio/audio.h +++ b/audio/audio.h @@ -26,18 +26,10 @@ #define QEMU_AUDIO_H #include "qemu/queue.h" +#include "qapi/qapi-types-audio.h" typedef void (*audio_callback_fn) (void *opaque, int avail); -typedef enum { - AUD_FMT_U8, - AUD_FMT_S8, - AUD_FMT_U16, - AUD_FMT_S16, - AUD_FMT_U32, - AUD_FMT_S32 -} audfmt_e; - #ifdef HOST_WORDS_BIGENDIAN #define AUDIO_HOST_ENDIANNESS 1 #else @@ -47,7 +39,7 @@ typedef enum { struct audsettings { int freq; int nchannels; - audfmt_e fmt; + AudioFormat fmt; int endianness; }; diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c index 6900008..b938fd6 100644 --- a/audio/audio_win_int.c +++ b/audio/audio_win_int.c @@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx, wfx->cbSize = 0; switch (as->fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: wfx->wBitsPerSample = 8; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: wfx->wBitsPerSample = 16; wfx->nAvgBytesPerSec <<= 1; wfx->nBlockAlign <<= 1; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: wfx->wBitsPerSample = 32; wfx->nAvgBytesPerSec <<= 2; wfx->nBlockAlign <<= 2; @@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx, switch (wfx->wBitsPerSample) { case 8: - as->fmt = AUD_FMT_U8; + as->fmt = AUDIO_FORMAT_U8; break; case 16: - as->fmt = AUD_FMT_S16; + as->fmt = AUDIO_FORMAT_S16; break; case 32: - as->fmt = AUD_FMT_S32; + as->fmt = AUDIO_FORMAT_S32; break; default: diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 6c69622..355e8fb 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -70,7 +70,7 @@ typedef struct OSSVoiceIn { struct oss_params { int freq; - audfmt_e fmt; + AudioFormat fmt; int nchannels; int nfrags; int fragsize; @@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len) return audio_pcm_sw_write (sw, buf, len); } -static int aud_to_ossfmt (audfmt_e fmt, int endianness) +static int aud_to_ossfmt (AudioFormat fmt, int endianness) { switch (fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return AFMT_S8; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return AFMT_U8; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: if (endianness) { return AFMT_S16_BE; } @@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness) return AFMT_S16_LE; } - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: if (endianness) { return AFMT_U16_BE; } @@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness) } } -static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness) +static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness) { switch (ossfmt) { case AFMT_S8: *endianness = 0; - *fmt = AUD_FMT_S8; + *fmt = AUDIO_FORMAT_S8; break; case AFMT_U8: *endianness = 0; - *fmt = AUD_FMT_U8; + *fmt = AUDIO_FORMAT_U8; break; case AFMT_S16_LE: *endianness = 0; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_LE: *endianness = 0; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case AFMT_S16_BE: *endianness = 1; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AFMT_U16_BE: *endianness = 1; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; default: @@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as, int endianness; int err; int fd; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; OSSConf *conf = drv_opaque; @@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) int endianness; int err; int fd; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; OSSConf *conf = drv_opaque; diff --git a/audio/paaudio.c b/audio/paaudio.c index 6153b90..8246f26 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len) return audio_pcm_sw_read (sw, buf, len); } -static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness) +static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness) { int format; switch (afmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: format = PA_SAMPLE_U8; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE; break; default: @@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness) return format; } -static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness) +static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness) { switch (fmt) { case PA_SAMPLE_U8: - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; case PA_SAMPLE_S16BE: *endianness = 1; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; case PA_SAMPLE_S16LE: *endianness = 0; - return AUD_FMT_S16; + return AUDIO_FORMAT_S16; case PA_SAMPLE_S32BE: *endianness = 1; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; case PA_SAMPLE_S32LE: *endianness = 0; - return AUD_FMT_S32; + return AUDIO_FORMAT_S32; default: dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt); - return AUD_FMT_U8; + return AUDIO_FORMAT_U8; } } diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c index f7ee70b..4cd4cba 100644 --- a/audio/sdlaudio.c +++ b/audio/sdlaudio.c @@ -68,19 +68,19 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...) AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ()); } -static int aud_to_sdlfmt (audfmt_e fmt) +static int aud_to_sdlfmt (AudioFormat fmt) { switch (fmt) { - case AUD_FMT_S8: + case AUDIO_FORMAT_S8: return AUDIO_S8; - case AUD_FMT_U8: + case AUDIO_FORMAT_U8: return AUDIO_U8; - case AUD_FMT_S16: + case AUDIO_FORMAT_S16: return AUDIO_S16LSB; - case AUD_FMT_U16: + case AUDIO_FORMAT_U16: return AUDIO_U16LSB; default: @@ -92,37 +92,37 @@ static int aud_to_sdlfmt (audfmt_e fmt) } } -static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness) +static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness) { switch (sdlfmt) { case AUDIO_S8: *endianness = 0; - *fmt = AUD_FMT_S8; + *fmt = AUDIO_FORMAT_S8; break; case AUDIO_U8: *endianness = 0; - *fmt = AUD_FMT_U8; + *fmt = AUDIO_FORMAT_U8; break; case AUDIO_S16LSB: *endianness = 0; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16LSB: *endianness = 0; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; case AUDIO_S16MSB: *endianness = 1; - *fmt = AUD_FMT_S16; + *fmt = AUDIO_FORMAT_S16; break; case AUDIO_U16MSB: *endianness = 1; - *fmt = AUD_FMT_U16; + *fmt = AUDIO_FORMAT_U16; break; default: @@ -265,7 +265,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as, SDL_AudioSpec req, obt; int endianness; int err; - audfmt_e effective_fmt; + AudioFormat effective_fmt; struct audsettings obt_as; req.freq = as->freq; diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c index 6ad0eaf..3aeb0cb 100644 --- a/audio/spiceaudio.c +++ b/audio/spiceaudio.c @@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as, settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ; #endif settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN; - settings.fmt = AUD_FMT_S16; + settings.fmt = AUDIO_FORMAT_S16; settings.endianness = AUDIO_HOST_ENDIANNESS; audio_pcm_init_info (&hw->info, &settings); @@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) settings.freq = SPICE_INTERFACE_RECORD_FREQ; #endif settings.nchannels = SPICE_INTERFACE_RECORD_CHAN; - settings.fmt = AUD_FMT_S16; + settings.fmt = AUDIO_FORMAT_S16; settings.endianness = AUDIO_HOST_ENDIANNESS; audio_pcm_init_info (&hw->info, &settings); diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 40adfa3..35a6147 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as, stereo = wav_as.nchannels == 2; switch (wav_as.fmt) { - case AUD_FMT_S8: - case AUD_FMT_U8: + case AUDIO_FORMAT_S8: + case AUDIO_FORMAT_U8: bits16 = 0; break; - case AUD_FMT_S16: - case AUD_FMT_U16: + case AUDIO_FORMAT_S16: + case AUDIO_FORMAT_U16: bits16 = 1; break; - case AUD_FMT_S32: - case AUD_FMT_U32: + case AUDIO_FORMAT_S32: + case AUDIO_FORMAT_U32: dolog ("WAVE files can not handle 32bit formats\n"); return -1; + + default: + abort(); } hdr[34] = bits16 ? 0x10 : 0x08; @@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...) static WAVConf glob_conf = { .settings.freq = 44100, .settings.nchannels = 2, - .settings.fmt = AUD_FMT_S16, + .settings.fmt = AUDIO_FORMAT_S16, .wav_path = "qemu.wav" }; diff --git a/audio/wavcapture.c b/audio/wavcapture.c index cd24570..74320df 100644 --- a/audio/wavcapture.c +++ b/audio/wavcapture.c @@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq, as.freq = freq; as.nchannels = 1 << stereo; - as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8; + as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8; as.endianness = 0; ops.notify = wav_notify; diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c index 94dffb2..4462239 100644 --- a/hw/arm/omap2.c +++ b/hw/arm/omap2.c @@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s) * does I2S specify it? */ /* All register writes are 16 bits so we we store 16-bit samples * in the buffers regardless of AGCFR[B8_16] value. */ - fmt.fmt = AUD_FMT_U16; + fmt.fmt = AUDIO_FORMAT_U16; s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice, "eac.codec.in", s, omap_eac_in_cb, &fmt); 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 */ } } diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c index cc0f9bc..11b09bd 100644 --- a/hw/display/xlnx_dp.c +++ b/hw/display/xlnx_dp.c @@ -1260,7 +1260,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp) as.freq = 44100; as.nchannels = 2; - as.fmt = AUD_FMT_S16; + as.fmt = AUDIO_FORMAT_S16; as.endianness = 0; AUD_register_card("xlnx_dp.audio", &s->aud_card); diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c index 2eb3cb9..4173161 100644 --- a/hw/input/tsc210x.c +++ b/hw/input/tsc210x.c @@ -318,7 +318,7 @@ static void tsc2102_audio_output_update(TSC210xState *s) fmt.endianness = 0; fmt.nchannels = 2; fmt.freq = s->codec.tx_rate; - fmt.fmt = AUD_FMT_S16; + fmt.fmt = AUDIO_FORMAT_S16; s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0], "tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt); diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c index 28ac7c5..c46d5ee 100644 --- a/hw/usb/dev-audio.c +++ b/hw/usb/dev-audio.c @@ -650,7 +650,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp) s->out.vol[1] = 240; /* 0 dB */ s->out.as.freq = USBAUDIO_SAMPLE_RATE; s->out.as.nchannels = 2; - s->out.as.fmt = AUD_FMT_S16; + s->out.as.fmt = AUDIO_FORMAT_S16; s->out.as.endianness = 0; streambuf_init(&s->out.buf, s->buffer); @@ -1013,16 +1013,16 @@ static void vnc_update_throttle_offset(VncState *vs) int bps; switch (vs->as.fmt) { default: - case AUD_FMT_U8: - case AUD_FMT_S8: + case AUDIO_FORMAT_U8: + case AUDIO_FORMAT_S8: bps = 1; break; - case AUD_FMT_U16: - case AUD_FMT_S16: + case AUDIO_FORMAT_U16: + case AUDIO_FORMAT_S16: bps = 2; break; - case AUD_FMT_U32: - case AUD_FMT_S32: + case AUDIO_FORMAT_U32: + case AUDIO_FORMAT_S32: bps = 4; break; } @@ -2369,12 +2369,12 @@ static int protocol_client_msg(VncState *vs, uint8_t *data, size_t len) if (len == 4) return 10; switch (read_u8(data, 4)) { - case 0: vs->as.fmt = AUD_FMT_U8; break; - case 1: vs->as.fmt = AUD_FMT_S8; break; - case 2: vs->as.fmt = AUD_FMT_U16; break; - case 3: vs->as.fmt = AUD_FMT_S16; break; - case 4: vs->as.fmt = AUD_FMT_U32; break; - case 5: vs->as.fmt = AUD_FMT_S32; break; + case 0: vs->as.fmt = AUDIO_FORMAT_U8; break; + case 1: vs->as.fmt = AUDIO_FORMAT_S8; break; + case 2: vs->as.fmt = AUDIO_FORMAT_U16; break; + case 3: vs->as.fmt = AUDIO_FORMAT_S16; break; + case 4: vs->as.fmt = AUDIO_FORMAT_U32; break; + case 5: vs->as.fmt = AUDIO_FORMAT_S32; break; default: VNC_DEBUG("Invalid audio format %d\n", read_u8(data, 4)); vnc_client_error(vs); @@ -3105,7 +3105,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc, vs->as.freq = 44100; vs->as.nchannels = 2; - vs->as.fmt = AUD_FMT_S16; + vs->as.fmt = AUDIO_FORMAT_S16; vs->as.endianness = 0; qemu_mutex_init(&vs->output_mutex); |