audio endianness API changes (malc)
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@2042 c046a42c-6fe2-441c-8c8c-71466251a162
This commit is contained in:
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219fb12503
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d929eba5d4
@ -662,12 +662,9 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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audio_pcm_init_info (
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&hw->info,
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&obt_as,
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audio_need_to_swap_endian (endianness)
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);
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
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@ -751,12 +748,9 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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audio_pcm_init_info (
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&hw->info,
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&obt_as,
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audio_need_to_swap_endian (endianness)
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);
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audio_pcm_init_info (&hw->info, &obt_as);
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hw->samples = obt.samples;
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alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
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@ -510,6 +510,18 @@ static void audio_print_settings (audsettings_t *as)
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AUD_log (NULL, "invalid(%d)", as->fmt);
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break;
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}
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AUD_log (NULL, "endianness=");
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switch (as->endianness) {
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case 0:
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AUD_log (NULL, "little");
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break;
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case 1:
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AUD_log (NULL, "big");
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break;
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default:
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AUD_log (NULL, "invalid");
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break;
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}
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AUD_log (NULL, "\n");
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}
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@ -518,6 +530,7 @@ static int audio_validate_settigs (audsettings_t *as)
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int invalid;
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invalid = as->nchannels != 1 && as->nchannels != 2;
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invalid |= as->endianness != 0 && as->endianness != 1;
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switch (as->fmt) {
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case AUD_FMT_S8:
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@ -531,11 +544,7 @@ static int audio_validate_settigs (audsettings_t *as)
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}
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invalid |= as->freq <= 0;
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if (invalid) {
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return -1;
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}
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return 0;
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return invalid ? -1 : 0;
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}
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static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
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@ -557,14 +566,11 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
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return info->freq == as->freq
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&& info->nchannels == as->nchannels
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&& info->sign == sign
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&& info->bits == bits;
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&& info->bits == bits
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&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
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}
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void audio_pcm_init_info (
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struct audio_pcm_info *info,
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audsettings_t *as,
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int swap_endian
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)
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void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
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{
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int bits = 8, sign = 0;
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@ -588,7 +594,7 @@ void audio_pcm_init_info (
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info->shift = (as->nchannels == 2) + (bits == 16);
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info->align = (1 << info->shift) - 1;
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info->bytes_per_second = info->freq << info->shift;
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info->swap_endian = swap_endian;
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info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
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}
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void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
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@ -610,7 +616,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
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int shift = info->nchannels - 1;
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short s = INT16_MAX;
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if (info->swap_endian) {
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if (info->swap_endianness) {
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s = bswap16 (s);
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}
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@ -635,16 +641,13 @@ static void noop_conv (st_sample_t *dst, const void *src,
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static CaptureVoiceOut *audio_pcm_capture_find_specific (
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AudioState *s,
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audsettings_t *as,
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int endian
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audsettings_t *as
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)
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{
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CaptureVoiceOut *cap;
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int swap_endian = audio_need_to_swap_endian (endian);
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for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
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if ((cap->hw.info.swap_endian == swap_endian)
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&& audio_pcm_info_eq (&cap->hw.info, as)) {
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if (audio_pcm_info_eq (&cap->hw.info, as)) {
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return cap;
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}
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}
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@ -1697,7 +1700,6 @@ AudioState *AUD_init (void)
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int AUD_add_capture (
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AudioState *s,
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audsettings_t *as,
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int endian,
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struct audio_capture_ops *ops,
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void *cb_opaque
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)
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@ -1725,7 +1727,7 @@ int AUD_add_capture (
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cb->ops = *ops;
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cb->opaque = cb_opaque;
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cap = audio_pcm_capture_find_specific (s, as, endian);
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cap = audio_pcm_capture_find_specific (s, as);
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if (cap) {
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LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
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return 0;
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@ -1755,7 +1757,7 @@ int AUD_add_capture (
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goto err2;
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}
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audio_pcm_init_info (&hw->info, as, endian);
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audio_pcm_init_info (&hw->info, as);
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cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
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if (!cap->buf) {
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@ -1768,7 +1770,7 @@ int AUD_add_capture (
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hw->clip = mixeng_clip
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[hw->info.nchannels == 2]
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[hw->info.sign]
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[hw->info.swap_endian]
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[hw->info.swap_endianness]
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[hw->info.bits == 16];
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LIST_INSERT_HEAD (&s->cap_head, cap, entries);
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@ -24,6 +24,7 @@
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#ifndef QEMU_AUDIO_H
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#define QEMU_AUDIO_H
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#include "config.h"
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#include "sys-queue.h"
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typedef void (*audio_callback_fn_t) (void *opaque, int avail);
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@ -35,10 +36,17 @@ typedef enum {
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AUD_FMT_S16
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} audfmt_e;
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#ifdef WORDS_BIGENDIAN
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#define AUDIO_HOST_ENDIANNESS 1
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#else
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#define AUDIO_HOST_ENDIANNESS 0
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#endif
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typedef struct {
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int freq;
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int nchannels;
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audfmt_e fmt;
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int endianness;
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} audsettings_t;
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struct audio_capture_ops {
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@ -74,7 +82,6 @@ void AUD_remove_card (QEMUSoundCard *card);
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int AUD_add_capture (
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AudioState *s,
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audsettings_t *as,
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int endian,
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struct audio_capture_ops *ops,
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void *opaque
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);
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@ -85,8 +92,7 @@ SWVoiceOut *AUD_open_out (
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const char *name,
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void *callback_opaque,
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audio_callback_fn_t callback_fn,
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audsettings_t *settings,
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int sw_endian
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audsettings_t *settings
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);
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void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
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@ -104,8 +110,7 @@ SWVoiceIn *AUD_open_in (
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const char *name,
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void *callback_opaque,
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audio_callback_fn_t callback_fn,
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audsettings_t *settings,
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int sw_endian
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audsettings_t *settings
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);
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void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
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@ -61,7 +61,7 @@ struct audio_pcm_info {
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int align;
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int shift;
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int bytes_per_second;
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int swap_endian;
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int swap_endianness;
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};
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typedef struct HWVoiceOut {
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@ -198,8 +198,7 @@ extern struct audio_driver coreaudio_audio_driver;
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extern struct audio_driver dsound_audio_driver;
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extern volume_t nominal_volume;
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void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as,
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int swap_endian);
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void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);
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void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
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int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
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@ -220,15 +219,6 @@ static inline int audio_ring_dist (int dst, int src, int len)
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return (dst >= src) ? (dst - src) : (len - src + dst);
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}
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static inline int audio_need_to_swap_endian (int endianness)
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{
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#ifdef WORDS_BIGENDIAN
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return endianness != 1;
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#else
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return endianness != 0;
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#endif
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}
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#if defined __GNUC__
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#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
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#define INIT_FIELD(f) . f
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@ -140,13 +140,12 @@ static int glue (audio_pcm_sw_init_, TYPE) (
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SW *sw,
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HW *hw,
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const char *name,
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audsettings_t *as,
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int endian
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audsettings_t *as
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)
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{
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int err;
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audio_pcm_init_info (&sw->info, as, audio_need_to_swap_endian (endian));
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audio_pcm_init_info (&sw->info, as);
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sw->hw = hw;
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sw->active = 0;
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#ifdef DAC
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@ -164,7 +163,7 @@ static int glue (audio_pcm_sw_init_, TYPE) (
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#endif
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[sw->info.nchannels == 2]
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[sw->info.sign]
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[sw->info.swap_endian]
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[sw->info.swap_endianness]
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[sw->info.bits == 16];
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sw->name = qemu_strdup (name);
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@ -288,7 +287,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
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#endif
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[hw->info.nchannels == 2]
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[hw->info.sign]
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[hw->info.swap_endian]
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[hw->info.swap_endianness]
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[hw->info.bits == 16];
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if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
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@ -336,8 +335,7 @@ static HW *glue (audio_pcm_hw_add_, TYPE) (AudioState *s, audsettings_t *as)
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static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
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AudioState *s,
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const char *sw_name,
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audsettings_t *as,
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int sw_endian
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audsettings_t *as
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)
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{
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SW *sw;
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@ -365,7 +363,7 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
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glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
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if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as, sw_endian)) {
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if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
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goto err3;
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}
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@ -407,8 +405,7 @@ SW *glue (AUD_open_, TYPE) (
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const char *name,
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void *callback_opaque ,
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audio_callback_fn_t callback_fn,
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audsettings_t *as,
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int sw_endian
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audsettings_t *as
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)
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{
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AudioState *s;
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@ -481,12 +478,12 @@ SW *glue (AUD_open_, TYPE) (
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}
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glue (audio_pcm_sw_fini_, TYPE) (sw);
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if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as, sw_endian)) {
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if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
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goto fail;
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}
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}
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else {
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sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as, sw_endian);
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sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as);
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if (!sw) {
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dolog ("Failed to create voice `%s'\n", name);
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return NULL;
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@ -295,7 +295,6 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
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UInt32 propertySize;
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int err;
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int bits = 8;
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int endianess = 0;
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const char *typ = "playback";
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AudioValueRange frameRange;
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@ -308,16 +307,9 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
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if (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) {
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bits = 16;
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endianess = 1;
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}
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audio_pcm_init_info (
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&hw->info,
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as,
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/* Following is irrelevant actually since we do not use
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mixengs clipping routines */
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audio_need_to_swap_endian (endianess)
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);
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audio_pcm_init_info (&hw->info, as);
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/* open default output device */
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propertySize = sizeof(core->outputDeviceID);
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@ -250,8 +250,8 @@ static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as)
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}
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ds->first_time = 1;
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audio_pcm_init_info (&hw->info, &obt_as, audio_need_to_swap_endian (0));
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obt_as.endianness = 0;
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audio_pcm_init_info (&hw->info, &obt_as);
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if (bc.dwBufferBytes & hw->info.align) {
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dolog (
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@ -358,6 +358,7 @@ static int fmod_init_out (HWVoiceOut *hw, audsettings_t *as)
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{
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int bits16, mode, channel;
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FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
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audsettings_t obt_as = *as;
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mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
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fmd->fmod_sample = FSOUND_Sample_Alloc (
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@ -384,7 +385,8 @@ static int fmod_init_out (HWVoiceOut *hw, audsettings_t *as)
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fmd->channel = channel;
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/* FMOD always operates on little endian frames? */
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audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
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obt_as.endianness = 0;
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audio_pcm_init_info (&hw->info, &obt_as);
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bits16 = (mode & FSOUND_16BITS) != 0;
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hw->samples = conf.nb_samples;
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return 0;
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@ -418,6 +420,7 @@ static int fmod_init_in (HWVoiceIn *hw, audsettings_t *as)
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{
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int bits16, mode;
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FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
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audsettings_t obt_as = *as;
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if (conf.broken_adc) {
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return -1;
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@ -440,7 +443,8 @@ static int fmod_init_in (HWVoiceIn *hw, audsettings_t *as)
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}
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/* FMOD always operates on little endian frames? */
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audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
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obt_as.endianness = 0;
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audio_pcm_init_info (&hw->info, &obt_as);
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bits16 = (mode & FSOUND_16BITS) != 0;
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hw->samples = conf.nb_samples;
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return 0;
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@ -68,7 +68,7 @@ static int no_write (SWVoiceOut *sw, void *buf, int len)
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static int no_init_out (HWVoiceOut *hw, audsettings_t *as)
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{
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audio_pcm_init_info (&hw->info, as, 0);
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audio_pcm_init_info (&hw->info, as);
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hw->samples = 1024;
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return 0;
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}
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@ -87,7 +87,7 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
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static int no_init_in (HWVoiceIn *hw, audsettings_t *as)
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{
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audio_pcm_init_info (&hw->info, as, 0);
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audio_pcm_init_info (&hw->info, as);
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hw->samples = 1024;
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return 0;
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}
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@ -453,12 +453,9 @@ static int oss_init_out (HWVoiceOut *hw, audsettings_t *as)
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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audio_pcm_init_info (
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&hw->info,
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&obt_as,
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audio_need_to_swap_endian (endianness)
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);
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audio_pcm_init_info (&hw->info, &obt_as);
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oss->nfrags = obt.nfrags;
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oss->fragsize = obt.fragsize;
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@ -597,12 +594,9 @@ static int oss_init_in (HWVoiceIn *hw, audsettings_t *as)
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obt_as.freq = obt.freq;
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obt_as.nchannels = obt.nchannels;
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obt_as.fmt = effective_fmt;
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obt_as.endianness = endianness;
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audio_pcm_init_info (
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&hw->info,
|
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&obt_as,
|
||||
audio_need_to_swap_endian (endianness)
|
||||
);
|
||||
audio_pcm_init_info (&hw->info, &obt_as);
|
||||
oss->nfrags = obt.nfrags;
|
||||
oss->fragsize = obt.fragsize;
|
||||
|
||||
|
@ -335,12 +335,9 @@ static int sdl_init_out (HWVoiceOut *hw, audsettings_t *as)
|
||||
obt_as.freq = obt.freq;
|
||||
obt_as.nchannels = obt.channels;
|
||||
obt_as.fmt = effective_fmt;
|
||||
obt_as.endianness = endianess;
|
||||
|
||||
audio_pcm_init_info (
|
||||
&hw->info,
|
||||
&obt_as,
|
||||
audio_need_to_swap_endian (endianess)
|
||||
);
|
||||
audio_pcm_init_info (&hw->info, &obt_as);
|
||||
hw->samples = obt.samples;
|
||||
|
||||
s->initialized = 1;
|
||||
|
@ -135,7 +135,8 @@ static int wav_init_out (HWVoiceOut *hw, audsettings_t *as)
|
||||
|
||||
hdr[34] = bits16 ? 0x10 : 0x08;
|
||||
|
||||
audio_pcm_init_info (&hw->info, &wav_as, audio_need_to_swap_endian (0));
|
||||
wav_as.endianness = 0;
|
||||
audio_pcm_init_info (&hw->info, &wav_as);
|
||||
|
||||
hw->samples = 1024;
|
||||
wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
|
||||
|
@ -70,6 +70,7 @@ void wav_capture (const char *path, int freq, int bits16, int stereo)
|
||||
as.freq = freq;
|
||||
as.nchannels = 1 << stereo;
|
||||
as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
|
||||
as.endianness = 0;
|
||||
|
||||
ops.state = wav_state_cb;
|
||||
ops.capture = wav_capture_cb;
|
||||
@ -97,5 +98,5 @@ void wav_capture (const char *path, int freq, int bits16, int stereo)
|
||||
}
|
||||
|
||||
qemu_put_buffer (wav->f, hdr, sizeof (hdr));
|
||||
AUD_add_capture (NULL, &as, 0, &ops, wav);
|
||||
AUD_add_capture (NULL, &as, &ops, wav);
|
||||
}
|
||||
|
@ -301,6 +301,7 @@ int Adlib_init (AudioState *audio)
|
||||
as.freq = conf.freq;
|
||||
as.nchannels = SHIFT;
|
||||
as.fmt = AUD_FMT_S16;
|
||||
as.endianness = AUDIO_HOST_ENDIANNESS;
|
||||
|
||||
AUD_register_card (audio, "adlib", &s->card);
|
||||
|
||||
@ -310,8 +311,7 @@ int Adlib_init (AudioState *audio)
|
||||
"adlib",
|
||||
s,
|
||||
adlib_callback,
|
||||
&as,
|
||||
0 /* XXX: little endian? */
|
||||
&as
|
||||
);
|
||||
if (!s->voice) {
|
||||
Adlib_fini (s);
|
||||
|
@ -423,6 +423,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
|
||||
as.freq = new_freq;
|
||||
as.nchannels = 1 << (new_fmt & 1);
|
||||
as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
|
||||
as.endianness = 0;
|
||||
|
||||
if (i == ADC_CHANNEL) {
|
||||
s->adc_voice =
|
||||
@ -432,8 +433,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
|
||||
"es1370.adc",
|
||||
s,
|
||||
es1370_adc_callback,
|
||||
&as,
|
||||
0 /* little endian */
|
||||
&as
|
||||
);
|
||||
}
|
||||
else {
|
||||
@ -444,8 +444,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
|
||||
i ? "es1370.dac2" : "es1370.dac1",
|
||||
s,
|
||||
i ? es1370_dac2_callback : es1370_dac1_callback,
|
||||
&as,
|
||||
0 /* litle endian */
|
||||
&as
|
||||
);
|
||||
}
|
||||
}
|
||||
|
@ -95,7 +95,7 @@ static void pcspk_callback(void *opaque, int free)
|
||||
int pcspk_audio_init(AudioState *audio)
|
||||
{
|
||||
PCSpkState *s = &pcspk_state;
|
||||
audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8};
|
||||
audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
|
||||
|
||||
if (!audio) {
|
||||
AUD_log(s_spk, "No audio state\n");
|
||||
@ -103,7 +103,7 @@ int pcspk_audio_init(AudioState *audio)
|
||||
}
|
||||
AUD_register_card(audio, s_spk, &s->card);
|
||||
|
||||
s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as, 0);
|
||||
s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as);
|
||||
if (!s->voice) {
|
||||
AUD_log(s_spk, "Could not open voice\n");
|
||||
return -1;
|
||||
|
16
hw/sb16.c
16
hw/sb16.c
@ -203,6 +203,7 @@ static void continue_dma8 (SB16State *s)
|
||||
as.freq = s->freq;
|
||||
as.nchannels = 1 << s->fmt_stereo;
|
||||
as.fmt = s->fmt;
|
||||
as.endianness = 0;
|
||||
|
||||
s->voice = AUD_open_out (
|
||||
&s->card,
|
||||
@ -210,8 +211,7 @@ static void continue_dma8 (SB16State *s)
|
||||
"sb16",
|
||||
s,
|
||||
SB_audio_callback,
|
||||
&as,
|
||||
0 /* little endian */
|
||||
&as
|
||||
);
|
||||
}
|
||||
|
||||
@ -348,6 +348,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
|
||||
as.freq = s->freq;
|
||||
as.nchannels = 1 << s->fmt_stereo;
|
||||
as.fmt = s->fmt;
|
||||
as.endianness = 0;
|
||||
|
||||
s->voice = AUD_open_out (
|
||||
&s->card,
|
||||
@ -355,8 +356,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
|
||||
"sb16",
|
||||
s,
|
||||
SB_audio_callback,
|
||||
&as,
|
||||
0 /* little endian */
|
||||
&as
|
||||
);
|
||||
}
|
||||
|
||||
@ -838,6 +838,7 @@ static void legacy_reset (SB16State *s)
|
||||
as.freq = s->freq;
|
||||
as.nchannels = 1;
|
||||
as.fmt = AUD_FMT_U8;
|
||||
as.endianness = 0;
|
||||
|
||||
s->voice = AUD_open_out (
|
||||
&s->card,
|
||||
@ -845,8 +846,7 @@ static void legacy_reset (SB16State *s)
|
||||
"sb16",
|
||||
s,
|
||||
SB_audio_callback,
|
||||
&as,
|
||||
0 /* little endian */
|
||||
&as
|
||||
);
|
||||
|
||||
/* Not sure about that... */
|
||||
@ -1371,6 +1371,7 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
|
||||
as.freq = s->freq;
|
||||
as.nchannels = 1 << s->fmt_stereo;
|
||||
as.fmt = s->fmt;
|
||||
as.endianness = 0;
|
||||
|
||||
s->voice = AUD_open_out (
|
||||
&s->card,
|
||||
@ -1378,8 +1379,7 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
|
||||
"sb16",
|
||||
s,
|
||||
SB_audio_callback,
|
||||
&as,
|
||||
0 /* little endian */
|
||||
&as
|
||||
);
|
||||
}
|
||||
|
||||
|
Loading…
Reference in New Issue
Block a user