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-rw-r--r--audio/alsaaudio.c5
-rw-r--r--audio/audio.c327
-rw-r--r--audio/audio.h14
-rw-r--r--audio/audio_int.h18
-rw-r--r--audio/audio_template.h12
-rw-r--r--audio/coreaudio.c2
-rw-r--r--audio/dsound_template.h14
-rw-r--r--audio/dsoundaudio.c8
-rw-r--r--audio/fmodaudio.c2
-rw-r--r--audio/noaudio.c34
-rw-r--r--audio/ossaudio.c26
-rw-r--r--audio/sdlaudio.c1
-rw-r--r--audio/wavaudio.c1
-rw-r--r--audio/wavcapture.c101
-rw-r--r--hw/es1370.c3
15 files changed, 514 insertions, 54 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 30f1e50..2cac396 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -61,8 +61,8 @@ static struct {
.size_in_usec_in = 1,
.size_in_usec_out = 1,
#endif
- .pcm_name_out = "hw:0,0",
- .pcm_name_in = "hw:0,0",
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
#ifdef HIGH_LATENCY
.buffer_size_in = 400000,
.period_size_in = 400000 / 4,
@@ -606,7 +606,6 @@ static int alsa_run_out (HWVoiceOut *hw)
}
}
- mixeng_clear (src, written);
rpos = (rpos + written) % hw->samples;
samples -= written;
len -= written;
diff --git a/audio/audio.c b/audio/audio.c
index f10025b..0de728c 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -29,6 +29,7 @@
/* #define DEBUG_PLIVE */
/* #define DEBUG_LIVE */
/* #define DEBUG_OUT */
+/* #define DEBUG_CAPTURE */
#define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
@@ -137,7 +138,7 @@ int audio_bug (const char *funcname, int cond)
if (cond) {
static int shown;
- AUD_log (NULL, "Error a bug that was just triggered in %s\n", funcname);
+ AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
@@ -621,6 +622,121 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
}
/*
+ * Capture
+ */
+static void noop_conv (st_sample_t *dst, const void *src,
+ int samples, volume_t *vol)
+{
+ (void) src;
+ (void) dst;
+ (void) samples;
+ (void) vol;
+}
+
+static CaptureVoiceOut *audio_pcm_capture_find_specific (
+ AudioState *s,
+ audsettings_t *as,
+ int endian
+ )
+{
+ CaptureVoiceOut *cap;
+ int swap_endian = audio_need_to_swap_endian (endian);
+
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ if ((cap->hw.info.swap_endian == swap_endian)
+ && audio_pcm_info_eq (&cap->hw.info, as)) {
+ return cap;
+ }
+ }
+ return NULL;
+}
+
+static void audio_notify_capture (CaptureVoiceOut *cap, int enabled)
+{
+ if (cap->hw.enabled != enabled) {
+ struct capture_callback *cb;
+
+ cap->hw.enabled = enabled;
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ cb->ops.state (cb->opaque, enabled);
+ }
+ }
+}
+
+static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
+{
+ HWVoiceOut *hw = &cap->hw;
+ SWVoiceOut *sw;
+ int enabled = 0;
+
+ for (sw = hw->sw_cap_head.lh_first; sw; sw = sw->cap_entries.le_next) {
+ if (sw->active) {
+ enabled = 1;
+ break;
+ }
+ }
+ audio_notify_capture (cap, enabled);
+}
+
+static void audio_detach_capture (HWVoiceOut *hw)
+{
+ SWVoiceOut *sw;
+
+ for (sw = hw->sw_cap_head.lh_first; sw; sw = sw->cap_entries.le_next) {
+ if (sw->rate) {
+ st_rate_stop (sw->rate);
+ sw->rate = NULL;
+ }
+
+ LIST_REMOVE (sw, entries);
+ LIST_REMOVE (sw, cap_entries);
+ qemu_free (sw);
+ audio_recalc_and_notify_capture ((CaptureVoiceOut *) sw->hw);
+ }
+}
+
+static int audio_attach_capture (AudioState *s, HWVoiceOut *hw)
+{
+ CaptureVoiceOut *cap;
+
+ audio_detach_capture (hw);
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ SWVoiceOut *sw;
+ HWVoiceOut *hw_cap;
+
+ hw_cap = &cap->hw;
+ sw = audio_calloc (AUDIO_FUNC, 1, sizeof (*sw));
+ if (!sw) {
+ dolog ("Could not allocate soft capture voice (%zu bytes)\n",
+ sizeof (*sw));
+ return -1;
+ }
+
+ sw->info = hw->info;
+ sw->hw = hw_cap;
+ sw->empty = 1;
+ sw->active = hw->enabled;
+ sw->conv = noop_conv;
+ sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
+ sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
+ if (!sw->rate) {
+ dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
+ qemu_free (sw);
+ return -1;
+ }
+ LIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
+ LIST_INSERT_HEAD (&hw->sw_cap_head, sw, cap_entries);
+ if (sw->active) {
+ audio_notify_capture (cap, 1);
+ }
+ else {
+ audio_recalc_and_notify_capture (cap);
+ }
+ }
+ return 0;
+}
+
+/*
* Hard voice (capture)
*/
static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
@@ -916,17 +1032,11 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
SWVoiceOut *temp_sw;
if (on) {
- int total;
-
hw->pending_disable = 0;
if (!hw->enabled) {
hw->enabled = 1;
hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
}
-
- if (sw->empty) {
- total = 0;
- }
}
else {
if (hw->enabled) {
@@ -940,6 +1050,13 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
hw->pending_disable = nb_active == 1;
}
}
+ for (temp_sw = hw->sw_cap_head.lh_first; temp_sw;
+ temp_sw = temp_sw->entries.le_next) {
+ temp_sw->active = hw->enabled;
+ if (hw->enabled) {
+ audio_notify_capture ((CaptureVoiceOut *) temp_sw->hw, 1);
+ }
+ }
sw->active = on;
}
}
@@ -1031,6 +1148,41 @@ static int audio_get_free (SWVoiceOut *sw)
return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}
+static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
+{
+ int n;
+
+ if (hw->enabled) {
+ SWVoiceOut *sw;
+
+ for (sw = hw->sw_cap_head.lh_first; sw; sw = sw->cap_entries.le_next) {
+ int rpos2 = rpos;
+
+ n = samples;
+ while (n) {
+ int till_end_of_hw = hw->samples - rpos2;
+ int to_write = audio_MIN (till_end_of_hw, n);
+ int bytes = to_write << hw->info.shift;
+ int written;
+
+ sw->buf = hw->mix_buf + rpos2;
+ written = audio_pcm_sw_write (sw, NULL, bytes);
+ if (written - bytes) {
+ dolog ("Could not mix %d bytes into a capture buffer",
+ bytes);
+ break;
+ }
+ n -= to_write;
+ rpos2 = (rpos2 + to_write) % hw->samples;
+ }
+ }
+ }
+
+ n = audio_MIN (samples, hw->samples - rpos);
+ mixeng_clear (hw->mix_buf + rpos, n);
+ mixeng_clear (hw->mix_buf, samples - n);
+}
+
static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
@@ -1038,7 +1190,7 @@ static void audio_run_out (AudioState *s)
while ((hw = audio_pcm_hw_find_any_enabled_out (s, hw))) {
int played;
- int live, free, nb_live, cleanup_required;
+ int live, free, nb_live, cleanup_required, prev_rpos;
live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
if (!nb_live) {
@@ -1057,6 +1209,11 @@ static void audio_run_out (AudioState *s)
hw->enabled = 0;
hw->pending_disable = 0;
hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+ for (sw = hw->sw_cap_head.lh_first; sw;
+ sw = sw->cap_entries.le_next) {
+ sw->active = 0;
+ audio_recalc_and_notify_capture ((CaptureVoiceOut *) sw->hw);
+ }
continue;
}
@@ -1072,6 +1229,7 @@ static void audio_run_out (AudioState *s)
continue;
}
+ prev_rpos = hw->rpos;
played = hw->pcm_ops->run_out (hw);
if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
@@ -1085,6 +1243,7 @@ static void audio_run_out (AudioState *s)
if (played) {
hw->ts_helper += played;
+ audio_capture_mix_and_clear (hw, prev_rpos, played);
}
cleanup_required = 0;
@@ -1158,12 +1317,60 @@ static void audio_run_in (AudioState *s)
}
}
+static void audio_run_capture (AudioState *s)
+{
+ CaptureVoiceOut *cap;
+
+ for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
+ int live, rpos, captured;
+ HWVoiceOut *hw = &cap->hw;
+ SWVoiceOut *sw;
+
+ captured = live = audio_pcm_hw_get_live_out (hw);
+ rpos = hw->rpos;
+ while (live) {
+ int left = hw->samples - rpos;
+ int to_capture = audio_MIN (live, left);
+ st_sample_t *src;
+ struct capture_callback *cb;
+
+ src = hw->mix_buf + rpos;
+ hw->clip (cap->buf, src, to_capture);
+ mixeng_clear (src, to_capture);
+
+ for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
+ cb->ops.capture (cb->opaque, cap->buf,
+ to_capture << hw->info.shift);
+ }
+ rpos = (rpos + to_capture) % hw->samples;
+ live -= to_capture;
+ }
+ hw->rpos = rpos;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (!sw->active && sw->empty) {
+ continue;
+ }
+
+ if (audio_bug (AUDIO_FUNC, captured > sw->total_hw_samples_mixed)) {
+ dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
+ captured, sw->total_hw_samples_mixed);
+ captured = sw->total_hw_samples_mixed;
+ }
+
+ sw->total_hw_samples_mixed -= captured;
+ sw->empty = sw->total_hw_samples_mixed == 0;
+ }
+ }
+}
+
static void audio_timer (void *opaque)
{
AudioState *s = opaque;
audio_run_out (s);
audio_run_in (s);
+ audio_run_capture (s);
qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
}
@@ -1327,8 +1534,14 @@ static void audio_atexit (void)
HWVoiceIn *hwi = NULL;
while ((hwo = audio_pcm_hw_find_any_enabled_out (s, hwo))) {
+ SWVoiceOut *sw;
+
hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
hwo->pcm_ops->fini_out (hwo);
+
+ for (sw = hwo->sw_cap_head.lh_first; sw; sw = sw->entries.le_next) {
+ audio_notify_capture ((CaptureVoiceOut *) sw->hw, 0);
+ }
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (s, hwi))) {
@@ -1383,6 +1596,7 @@ AudioState *AUD_init (void)
LIST_INIT (&s->hw_head_out);
LIST_INIT (&s->hw_head_in);
+ LIST_INIT (&s->cap_head);
atexit (audio_atexit);
s->ts = qemu_new_timer (vm_clock, audio_timer, s);
@@ -1479,3 +1693,100 @@ AudioState *AUD_init (void)
qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + conf.period.ticks);
return s;
}
+
+int AUD_add_capture (
+ AudioState *s,
+ audsettings_t *as,
+ int endian,
+ struct audio_capture_ops *ops,
+ void *cb_opaque
+ )
+{
+ CaptureVoiceOut *cap;
+ struct capture_callback *cb;
+
+ if (!s) {
+ /* XXX suppress */
+ s = &glob_audio_state;
+ }
+
+ if (audio_validate_settigs (as)) {
+ dolog ("Invalid settings were passed when trying to add capture\n");
+ audio_print_settings (as);
+ return -1;
+ }
+
+ cb = audio_calloc (AUDIO_FUNC, 1, sizeof (*cb));
+ if (!cb) {
+ dolog ("Could not allocate capture callback information, size %zu\n",
+ sizeof (*cb));
+ goto err0;
+ }
+ cb->ops = *ops;
+ cb->opaque = cb_opaque;
+
+ cap = audio_pcm_capture_find_specific (s, as, endian);
+ if (cap) {
+ LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+ return 0;
+ }
+ else {
+ HWVoiceOut *hw;
+ CaptureVoiceOut *cap;
+
+ cap = audio_calloc (AUDIO_FUNC, 1, sizeof (*cap));
+ if (!cap) {
+ dolog ("Could not allocate capture voice, size %zu\n",
+ sizeof (*cap));
+ goto err1;
+ }
+
+ hw = &cap->hw;
+ LIST_INIT (&hw->sw_head);
+ LIST_INIT (&cap->cb_head);
+
+ /* XXX find a more elegant way */
+ hw->samples = 4096 * 4;
+ hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
+ sizeof (st_sample_t));
+ if (!hw->mix_buf) {
+ dolog ("Could not allocate capture mix buffer (%d samples)\n",
+ hw->samples);
+ goto err2;
+ }
+
+ audio_pcm_init_info (&hw->info, as, endian);
+
+ cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+ if (!cap->buf) {
+ dolog ("Could not allocate capture buffer "
+ "(%d samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
+ goto err3;
+ }
+
+ hw->clip = mixeng_clip
+ [hw->info.nchannels == 2]
+ [hw->info.sign]
+ [hw->info.swap_endian]
+ [hw->info.bits == 16];
+
+ LIST_INSERT_HEAD (&s->cap_head, cap, entries);
+ LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
+
+ hw = NULL;
+ while ((hw = audio_pcm_hw_find_any_out (s, hw))) {
+ audio_attach_capture (s, hw);
+ }
+ return 0;
+
+ err3:
+ qemu_free (cap->hw.mix_buf);
+ err2:
+ qemu_free (cap);
+ err1:
+ qemu_free (cb);
+ err0:
+ return -1;
+ }
+}
diff --git a/audio/audio.h b/audio/audio.h
index 169b5f6..4e1a694 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -41,6 +41,11 @@ typedef struct {
audfmt_e fmt;
} audsettings_t;
+struct audio_capture_ops {
+ void (*state) (void *opaque, int enabled);
+ void (*capture) (void *opaque, void *buf, int size);
+};
+
typedef struct AudioState AudioState;
typedef struct SWVoiceOut SWVoiceOut;
typedef struct SWVoiceIn SWVoiceIn;
@@ -66,6 +71,13 @@ AudioState *AUD_init (void);
void AUD_help (void);
void AUD_register_card (AudioState *s, const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
+int AUD_add_capture (
+ AudioState *s,
+ audsettings_t *as,
+ int endian,
+ struct audio_capture_ops *ops,
+ void *opaque
+ );
SWVoiceOut *AUD_open_out (
QEMUSoundCard *card,
@@ -111,7 +123,7 @@ static inline void *advance (void *p, int incr)
}
uint32_t popcount (uint32_t u);
-inline uint32_t lsbindex (uint32_t u);
+uint32_t lsbindex (uint32_t u);
#ifdef __GNUC__
#define audio_MIN(a, b) ( __extension__ ({ \
diff --git a/audio/audio_int.h b/audio/audio_int.h
index ca240cc..c01c16a 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -79,6 +79,7 @@ typedef struct HWVoiceOut {
int samples;
LIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
+ LIST_HEAD (sw_cap_listhead, SWVoiceOut) sw_cap_head;
struct audio_pcm_ops *pcm_ops;
LIST_ENTRY (HWVoiceOut) entries;
} HWVoiceOut;
@@ -115,6 +116,7 @@ struct SWVoiceOut {
volume_t vol;
struct audio_callback callback;
LIST_ENTRY (SWVoiceOut) entries;
+ LIST_ENTRY (SWVoiceOut) cap_entries;
};
struct SWVoiceIn {
@@ -160,14 +162,28 @@ struct audio_pcm_ops {
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
+struct capture_callback {
+ struct audio_capture_ops ops;
+ void *opaque;
+ LIST_ENTRY (capture_callback) entries;
+};
+
+typedef struct CaptureVoiceOut {
+ HWVoiceOut hw;
+ void *buf;
+ LIST_HEAD (cb_listhead, capture_callback) cb_head;
+ LIST_ENTRY (CaptureVoiceOut) entries;
+} CaptureVoiceOut;
+
struct AudioState {
struct audio_driver *drv;
void *drv_opaque;
QEMUTimer *ts;
- LIST_HEAD (card_head, QEMUSoundCard) card_head;
+ LIST_HEAD (card_listhead, QEMUSoundCard) card_head;
LIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
LIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
+ LIST_HEAD (cap_listhead, CaptureVoiceOut) cap_head;
int nb_hw_voices_out;
int nb_hw_voices_in;
};
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 419a4aa..04b3023 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -200,6 +200,9 @@ static void glue (audio_pcm_hw_gc_, TYPE) (AudioState *s, HW **hwp)
HW *hw = *hwp;
if (!hw->sw_head.lh_first) {
+#ifdef DAC
+ audio_detach_capture (hw);
+#endif
LIST_REMOVE (hw, entries);
glue (s->nb_hw_voices_, TYPE) += 1;
glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
@@ -266,7 +269,9 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
hw->pcm_ops = drv->pcm_ops;
LIST_INIT (&hw->sw_head);
-
+#ifdef DAC
+ LIST_INIT (&hw->sw_cap_head);
+#endif
if (glue (hw->pcm_ops->init_, TYPE) (hw, as)) {
goto err0;
}
@@ -292,6 +297,9 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
LIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
glue (s->nb_hw_voices_, TYPE) -= 1;
+#ifdef DAC
+ audio_attach_capture (s, hw);
+#endif
return hw;
err1:
@@ -542,7 +550,7 @@ uint64_t glue (AUD_get_elapsed_usec_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
cur_ts = sw->hw->ts_helper;
old_ts = ts->old_ts;
- /* dolog ("cur %" PRId64 " old %" PRId64 "\n", cur_ts, old_ts); */
+ /* dolog ("cur %lld old %lld\n", cur_ts, old_ts); */
if (cur_ts >= old_ts) {
delta = cur_ts - old_ts;
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 534fb3e..34e416d 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -275,8 +275,6 @@ static OSStatus audioDeviceIOProc(
#endif
}
- /* cleanup */
- mixeng_clear (src, frameCount);
rpos = (rpos + frameCount) % hw->samples;
core->decr += frameCount;
core->rpos = rpos;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 38ba5b9..96f7cc7 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -70,7 +70,13 @@ static int glue (dsound_lock_, TYPE) (
int i;
LPVOID p1 = NULL, p2 = NULL;
DWORD blen1 = 0, blen2 = 0;
+ DWORD flag;
+#ifdef DSBTYPE_IN
+ flag = entire ? DSCBLOCK_ENTIREBUFFER : 0;
+#else
+ flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
+#endif
for (i = 0; i < conf.lock_retries; ++i) {
hr = glue (IFACE, _Lock) (
buf,
@@ -80,13 +86,7 @@ static int glue (dsound_lock_, TYPE) (
&blen1,
&p2,
&blen2,
- (entire
-#ifdef DSBTYPE_IN
- ? DSCBLOCK_ENTIREBUFFER
-#else
- ? DSBLOCK_ENTIREBUFFER
-#endif
- : 0)
+ flag
);
if (FAILED (hr)) {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 63c5a50..90a0333 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -453,13 +453,11 @@ static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
if (src_len1) {
hw->clip (dst, src1, src_len1);
- mixeng_clear (src1, src_len1);
}
if (src_len2) {
dst = advance (dst, src_len1 << hw->info.shift);
hw->clip (dst, src2, src_len2);
- mixeng_clear (src2, src_len2);
}
hw->rpos = pos % hw->samples;
@@ -987,6 +985,12 @@ static void *dsound_audio_init (void)
hr = IDirectSound_Initialize (s->dsound, NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize DirectSound\n");
+
+ hr = IDirectSound_Release (s->dsound);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Could not release DirectSound\n");
+ }
+ s->dsound = NULL;
return NULL;
}
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 072d8a8..23f2677 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -153,13 +153,11 @@ static void fmod_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
if (src_len1) {
hw->clip (dst, src1, src_len1);
- mixeng_clear (src1, src_len1);
}
if (src_len2) {
dst = advance (dst, src_len1 << hw->info.shift);
hw->clip (dst, src2, src_len2);
- mixeng_clear (src2, src_len2);
}
hw->rpos = pos % hw->samples;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index aa35811..314f617 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -40,22 +40,21 @@ static int no_run_out (HWVoiceOut *hw)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
int live, decr, samples;
- int64_t now = qemu_get_clock (vm_clock);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
-
- if (bytes > INT_MAX) {
- samples = INT_MAX >> hw->info.shift;
- }
- else {
- samples = bytes >> hw->info.shift;
- }
+ int64_t now;
+ int64_t ticks;
+ int64_t bytes;
live = audio_pcm_hw_get_live_out (&no->hw);
if (!live) {
return 0;
}
+ now = qemu_get_clock (vm_clock);
+ ticks = now - no->old_ticks;
+ bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
+ bytes = audio_MIN (bytes, INT_MAX);
+ samples = bytes >> hw->info.shift;
+
no->old_ticks = now;
decr = audio_MIN (live, samples);
hw->rpos = (hw->rpos + decr) % hw->samples;
@@ -101,17 +100,20 @@ static void no_fini_in (HWVoiceIn *hw)
static int no_run_in (HWVoiceIn *hw)
{
NoVoiceIn *no = (NoVoiceIn *) hw;
- int64_t now = qemu_get_clock (vm_clock);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
int live = audio_pcm_hw_get_live_in (hw);
int dead = hw->samples - live;
int samples;
- bytes = audio_MIN (bytes, INT_MAX);
- samples = bytes >> hw->info.shift;
- samples = audio_MIN (samples, dead);
+ if (dead) {
+ int64_t now = qemu_get_clock (vm_clock);
+ int64_t ticks = now - no->old_ticks;
+ int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
+ no->old_ticks = now;
+ bytes = audio_MIN (bytes, INT_MAX);
+ samples = bytes >> hw->info.shift;
+ samples = audio_MIN (samples, dead);
+ }
return samples;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 7d12f9e..0bdc8ea 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -55,12 +55,14 @@ static struct {
int fragsize;
const char *devpath_out;
const char *devpath_in;
+ int debug;
} conf = {
.try_mmap = 0,
.nfrags = 4,
.fragsize = 4096,
.devpath_out = "/dev/dsp",
- .devpath_in = "/dev/dsp"
+ .devpath_in = "/dev/dsp",
+ .debug = 0
};
struct oss_params {
@@ -324,9 +326,20 @@ static int oss_run_out (HWVoiceOut *hw)
return 0;
}
- if (abinfo.bytes < 0 || abinfo.bytes > bufsize) {
- ldebug ("warning: Invalid available size, size=%d bufsize=%d\n",
- abinfo.bytes, bufsize);
+ if (abinfo.bytes > bufsize) {
+ if (conf.debug) {
+ dolog ("warning: Invalid available size, size=%d bufsize=%d\n"
+ "please report your OS/audio hw to malc@pulsesoft.com\n",
+ abinfo.bytes, bufsize);
+ }
+ abinfo.bytes = bufsize;
+ }
+
+ if (abinfo.bytes < 0) {
+ if (conf.debug) {
+ dolog ("warning: Invalid available size, size=%d bufsize=%d\n",
+ abinfo.bytes, bufsize);
+ }
return 0;
}
@@ -369,15 +382,12 @@ static int oss_run_out (HWVoiceOut *hw)
"alignment %d\n",
wbytes, written, hw->info.align + 1);
}
- mixeng_clear (src, wsamples);
decr -= wsamples;
rpos = (rpos + wsamples) % hw->samples;
break;
}
}
- mixeng_clear (src, convert_samples);
-
rpos = (rpos + convert_samples) % hw->samples;
samples -= convert_samples;
}
@@ -730,6 +740,8 @@ static struct audio_option oss_options[] = {
"Path to DAC device", NULL, 0},
{"ADC_DEV", AUD_OPT_STR, &conf.devpath_in,
"Path to ADC device", NULL, 0},
+ {"DEBUG", AUD_OPT_BOOL, &conf.debug,
+ "Turn on some debugging messages", NULL, 0},
{NULL, 0, NULL, NULL, NULL, 0}
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 713c784..9fe2128 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -240,7 +240,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
hw->clip (buf, src, chunk);
- mixeng_clear (src, chunk);
sdl->rpos = (sdl->rpos + chunk) % hw->samples;
to_mix -= chunk;
buf += chunk << hw->info.shift;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 18d2bb0..ca1e99f 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -81,7 +81,6 @@ static int wav_run_out (HWVoiceOut *hw)
hw->clip (dst, src, convert_samples);
qemu_put_buffer (wav->f, dst, convert_samples << hw->info.shift);
- mixeng_clear (src, convert_samples);
rpos = (rpos + convert_samples) % hw->samples;
samples -= convert_samples;
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
new file mode 100644
index 0000000..33f04c5
--- /dev/null
+++ b/audio/wavcapture.c
@@ -0,0 +1,101 @@
+#include "vl.h"
+
+typedef struct {
+ QEMUFile *f;
+ int bytes;
+} WAVState;
+
+/* VICE code: Store number as little endian. */
+static void le_store (uint8_t *buf, uint32_t val, int len)
+{
+ int i;
+ for (i = 0; i < len; i++) {
+ buf[i] = (uint8_t) (val & 0xff);
+ val >>= 8;
+ }
+}
+
+static void wav_state_cb (void *opaque, int enabled)
+{
+ WAVState *wav = opaque;
+
+ if (!enabled) {
+ uint8_t rlen[4];
+ uint8_t dlen[4];
+ uint32_t datalen = wav->bytes;
+ uint32_t rifflen = datalen + 36;
+
+ if (!wav->f) {
+ return;
+ }
+
+ le_store (rlen, rifflen, 4);
+ le_store (dlen, datalen, 4);
+
+ qemu_fseek (wav->f, 4, SEEK_SET);
+ qemu_put_buffer (wav->f, rlen, 4);
+
+ qemu_fseek (wav->f, 32, SEEK_CUR);
+ qemu_put_buffer (wav->f, dlen, 4);
+ }
+ else {
+ qemu_fseek (wav->f, 0, SEEK_END);
+ }
+}
+
+static void wav_capture_cb (void *opaque, void *buf, int size)
+{
+ WAVState *wav = opaque;
+
+ qemu_put_buffer (wav->f, buf, size);
+ wav->bytes += size;
+}
+
+void wav_capture (const char *path, int freq, int bits16, int stereo)
+{
+ WAVState *wav;
+ uint8_t hdr[] = {
+ 0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
+ 0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
+ 0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
+ 0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
+ };
+ audsettings_t as;
+ struct audio_capture_ops ops;
+ int shift;
+
+ stereo = !!stereo;
+ bits16 = !!bits16;
+
+ as.freq = freq;
+ as.nchannels = 1 << stereo;
+ as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+
+ ops.state = wav_state_cb;
+ ops.capture = wav_capture_cb;
+
+ wav = qemu_mallocz (sizeof (*wav));
+ if (!wav) {
+ AUD_log ("wav", "Could not allocate memory (%zu bytes)", sizeof (*wav));
+ return;
+ }
+
+ shift = bits16 + stereo;
+ hdr[34] = bits16 ? 0x10 : 0x08;
+
+ le_store (hdr + 22, as.nchannels, 2);
+ le_store (hdr + 24, freq, 4);
+ le_store (hdr + 28, freq << shift, 4);
+ le_store (hdr + 32, 1 << shift, 2);
+
+ wav->f = fopen (path, "wb");
+ if (!wav->f) {
+ AUD_log ("wav", "Failed to open wave file `%s'\nReason: %s\n",
+ path, strerror (errno));
+ qemu_free (wav);
+ return;
+ }
+
+ qemu_put_buffer (wav->f, hdr, sizeof (hdr));
+ AUD_add_capture (NULL, &as, 0, &ops, wav);
+}
diff --git a/hw/es1370.c b/hw/es1370.c
index 9fddd9d..2aa2db9 100644
--- a/hw/es1370.c
+++ b/hw/es1370.c
@@ -479,9 +479,10 @@ static inline uint32_t es1370_fixup (ES1370State *s, uint32_t addr)
IO_WRITE_PROTO (es1370_writeb)
{
ES1370State *s = opaque;
- addr = es1370_fixup (s, addr);
uint32_t shift, mask;
+ addr = es1370_fixup (s, addr);
+
switch (addr) {
case ES1370_REG_CONTROL:
case ES1370_REG_CONTROL + 1:
generated by cgit v1.1 at 2024-12-25 03:23:53 +0000