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[Qemu-devel] [PATCH v3 13/14] audio: use size_t where makes sense
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From: |
Kővágó, Zoltán |
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Subject: |
[Qemu-devel] [PATCH v3 13/14] audio: use size_t where makes sense |
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Date: |
Sun, 4 Aug 2019 19:04:25 +0200 |
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
audio/audio.h | 4 +-
audio/audio_int.h | 26 +++----
audio/audio_template.h | 14 ++--
audio/mixeng.h | 9 +--
audio/rate_template.h | 2 +-
include/sysemu/replay.h | 4 +-
audio/alsaaudio.c | 26 +++----
audio/audio.c | 156 ++++++++++++++++++++--------------------
audio/coreaudio.c | 10 +--
audio/dsoundaudio.c | 11 +--
audio/noaudio.c | 16 ++---
audio/ossaudio.c | 45 ++++++------
audio/paaudio.c | 44 ++++++------
audio/sdlaudio.c | 20 +++---
audio/spiceaudio.c | 12 ++--
audio/wavaudio.c | 8 +--
replay/replay-audio.c | 16 ++---
replay/replay.c | 2 +-
18 files changed, 212 insertions(+), 213 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index 4a95758516..2db27bba7b 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -112,7 +112,7 @@ SWVoiceOut *AUD_open_out (
);
void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
-int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+size_t AUD_write (SWVoiceOut *sw, void *pcm_buf, size_t size);
int AUD_get_buffer_size_out (SWVoiceOut *sw);
void AUD_set_active_out (SWVoiceOut *sw, int on);
int AUD_is_active_out (SWVoiceOut *sw);
@@ -133,7 +133,7 @@ SWVoiceIn *AUD_open_in (
);
void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
-int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+size_t AUD_read (SWVoiceIn *sw, void *pcm_buf, size_t size);
void AUD_set_active_in (SWVoiceIn *sw, int on);
int AUD_is_active_in (SWVoiceIn *sw);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 003b7ab8cc..a674c5374a 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -61,12 +61,12 @@ typedef struct HWVoiceOut {
f_sample *clip;
- int rpos;
+ size_t rpos;
uint64_t ts_helper;
struct st_sample *mix_buf;
- int samples;
+ size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
int ctl_caps;
@@ -82,13 +82,13 @@ typedef struct HWVoiceIn {
t_sample *conv;
- int wpos;
- int total_samples_captured;
+ size_t wpos;
+ size_t total_samples_captured;
uint64_t ts_helper;
struct st_sample *conv_buf;
- int samples;
+ size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
int ctl_caps;
struct audio_pcm_ops *pcm_ops;
@@ -103,7 +103,7 @@ struct SWVoiceOut {
int64_t ratio;
struct st_sample *buf;
void *rate;
- int total_hw_samples_mixed;
+ size_t total_hw_samples_mixed;
int active;
int empty;
HWVoiceOut *hw;
@@ -120,7 +120,7 @@ struct SWVoiceIn {
struct audio_pcm_info info;
int64_t ratio;
void *rate;
- int total_hw_samples_acquired;
+ size_t total_hw_samples_acquired;
struct st_sample *buf;
f_sample *clip;
HWVoiceIn *hw;
@@ -149,12 +149,12 @@ struct audio_driver {
struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
- int (*run_out) (HWVoiceOut *hw, int live);
+ size_t (*run_out)(HWVoiceOut *hw, size_t live);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
- int (*run_in) (HWVoiceIn *hw);
+ size_t (*run_in)(HWVoiceIn *hw);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
@@ -208,10 +208,10 @@ audio_driver *audio_driver_lookup(const char *name);
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int
len);
-int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
+size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
-int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
- int live, int pending);
+size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
+ size_t live, size_t pending);
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
@@ -224,7 +224,7 @@ void audio_run(AudioState *s, const char *msg);
#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
-static inline int audio_ring_dist (int dst, int src, int len)
+static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
{
return (dst >= src) ? (dst - src) : (len - src + dst);
}
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 54f07338e7..2562bf5f00 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -75,16 +75,16 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW
*hw)
HWBUF = NULL;
}
-static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
+static bool glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
if (!HWBUF) {
- dolog ("Could not allocate " NAME " buffer (%d samples)\n",
- hw->samples);
- return -1;
+ dolog("Could not allocate " NAME " buffer (%zu samples)\n",
+ hw->samples);
+ return false;
}
- return 0;
+ return true;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -265,7 +265,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
}
if (audio_bug(__func__, hw->samples <= 0)) {
- dolog ("hw->samples=%d\n", hw->samples);
+ dolog("hw->samples=%zd\n", hw->samples);
goto err1;
}
@@ -279,7 +279,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
- if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
+ if (!glue(audio_pcm_hw_alloc_resources_, TYPE)(hw)) {
goto err1;
}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index b53a5ef99a..18e62c7c49 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -33,6 +33,7 @@ struct st_sample { mixeng_real l; mixeng_real r; };
struct mixeng_volume { int mute; int64_t r; int64_t l; };
struct st_sample { int64_t l; int64_t r; };
#endif
+typedef struct st_sample st_sample;
typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
@@ -41,10 +42,10 @@ extern t_sample *mixeng_conv[2][2][2][3];
extern f_sample *mixeng_clip[2][2][2][3];
void *st_rate_start (int inrate, int outrate);
-void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample
*obuf,
- int *isamp, int *osamp);
-void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample
*obuf,
- int *isamp, int *osamp);
+void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
+ size_t *isamp, size_t *osamp);
+void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
+ size_t *isamp, size_t *osamp);
void st_rate_stop (void *opaque);
void mixeng_clear (struct st_sample *buf, int len);
void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
diff --git a/audio/rate_template.h b/audio/rate_template.h
index 6e93588877..f94c940c61 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -28,7 +28,7 @@
* Return number of samples processed.
*/
void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
- int *isamp, int *osamp)
+ size_t *isamp, size_t *osamp)
{
struct rate *rate = opaque;
struct st_sample *istart, *iend;
diff --git a/include/sysemu/replay.h b/include/sysemu/replay.h
index 3a7c58e423..5c0a91e44f 100644
--- a/include/sysemu/replay.h
+++ b/include/sysemu/replay.h
@@ -179,9 +179,9 @@ void replay_net_packet_event(ReplayNetState *rns, unsigned
flags,
/* Audio */
/*! Saves/restores number of played samples of audio out operation. */
-void replay_audio_out(int *played);
+void replay_audio_out(size_t *played);
/*! Saves/restores recorded samples of audio in operation. */
-void replay_audio_in(int *recorded, void *samples, int *wpos, int size);
+void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t
size);
/* VM state operations */
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index e9e3a4819c..591344dccd 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -681,10 +681,10 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
}
}
-static int alsa_run_out (HWVoiceOut *hw, int live)
+static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- int decr;
+ size_t decr;
snd_pcm_sframes_t avail;
avail = alsa_get_avail (alsa->handle);
@@ -739,8 +739,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as,
alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}
@@ -841,8 +841,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}
@@ -863,17 +863,17 @@ static void alsa_fini_in (HWVoiceIn *hw)
alsa->pcm_buf = NULL;
}
-static int alsa_run_in (HWVoiceIn *hw)
+static size_t alsa_run_in(HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
int hwshift = hw->info.shift;
int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int decr;
+ size_t live = audio_pcm_hw_get_live_in (hw);
+ size_t dead = hw->samples - live;
+ size_t decr;
struct {
- int add;
- int len;
+ size_t add;
+ size_t len;
} bufs[2] = {
{ .add = hw->wpos, .len = 0 },
{ .add = 0, .len = 0 }
@@ -913,7 +913,7 @@ static int alsa_run_in (HWVoiceIn *hw)
}
}
- decr = MIN (dead, avail);
+ decr = MIN(dead, avail);
if (!decr) {
return 0;
}
diff --git a/audio/audio.c b/audio/audio.c
index 254499dfa1..d3c639211d 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -526,10 +526,10 @@ static int audio_attach_capture (HWVoiceOut *hw)
/*
* Hard voice (capture)
*/
-static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
- int m = hw->total_samples_captured;
+ size_t m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
@@ -539,28 +539,28 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
return m;
}
-int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
+size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
- int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
}
-int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
- int live, int pending)
+size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
+ size_t live, size_t pending)
{
- int left = hw->samples - pending;
- int len = MIN (left, live);
- int clipped = 0;
+ size_t left = hw->samples - pending;
+ size_t len = MIN (left, live);
+ size_t clipped = 0;
while (len) {
struct st_sample *src = hw->mix_buf + hw->rpos;
uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
- int samples_till_end_of_buf = hw->samples - hw->rpos;
- int samples_to_clip = MIN (len, samples_till_end_of_buf);
+ size_t samples_till_end_of_buf = hw->samples - hw->rpos;
+ size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
@@ -574,14 +574,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
/*
* Soft voice (capture)
*/
-static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
+static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
- int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- int rpos;
+ ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ ssize_t rpos;
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
@@ -594,17 +594,17 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
}
}
-static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
+static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
{
HWVoiceIn *hw = sw->hw;
- int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+ size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
@@ -618,9 +618,9 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int
size)
while (swlim) {
src = hw->conv_buf + rpos;
- isamp = hw->wpos - rpos;
- /* XXX: <= ? */
- if (isamp <= 0) {
+ if (hw->wpos > rpos) {
+ isamp = hw->wpos - rpos;
+ } else {
isamp = hw->samples - rpos;
}
@@ -629,11 +629,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int
size)
}
osamp = swlim;
- if (audio_bug(__func__, osamp < 0)) {
- dolog ("osamp=%d\n", osamp);
- return 0;
- }
-
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->samples;
@@ -654,10 +649,10 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf,
int size)
/*
* Hard voice (playback)
*/
-static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
+static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
- int m = INT_MAX;
+ size_t m = SIZE_MAX;
int nb_live = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
@@ -671,9 +666,9 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int
*nb_livep)
return m;
}
-static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
+static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
{
- int smin;
+ size_t smin;
int nb_live1;
smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
@@ -682,10 +677,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int
*nb_live)
}
if (nb_live1) {
- int live = smin;
+ size_t live = smin;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
@@ -696,10 +691,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int
*nb_live)
/*
* Soft voice (playback)
*/
-static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf, int size)
+static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
{
- int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
- int ret = 0, pos = 0, total = 0;
+ size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim,
blck;
+ size_t ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
@@ -708,8 +703,8 @@ static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf,
int size)
hwsamples = sw->hw->samples;
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live < 0 || live > hwsamples)) {
- dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ if (audio_bug(__func__, live > hwsamples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
return 0;
}
@@ -763,7 +758,7 @@ static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf,
int size)
#ifdef DEBUG_OUT
dolog (
- "%s: write size %d ret %d total sw %d\n",
+ "%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
size >> sw->info.shift,
ret,
@@ -842,7 +837,7 @@ static void audio_timer (void *opaque)
/*
* Public API
*/
-int AUD_write (SWVoiceOut *sw, void *buf, int size)
+size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
if (!sw) {
/* XXX: Consider options */
@@ -857,7 +852,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
return audio_pcm_sw_write(sw, buf, size);
}
-int AUD_read (SWVoiceIn *sw, void *buf, int size)
+size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
if (!sw) {
/* XXX: Consider options */
@@ -966,17 +961,17 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
}
}
-static int audio_get_avail (SWVoiceIn *sw)
+static size_t audio_get_avail (SWVoiceIn *sw)
{
- int live;
+ size_t live;
if (!sw) {
return 0;
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->samples)) {
+ dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -989,9 +984,9 @@ static int audio_get_avail (SWVoiceIn *sw)
return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}
-static int audio_get_free (SWVoiceOut *sw)
+static size_t audio_get_free(SWVoiceOut *sw)
{
- int live, dead;
+ size_t live, dead;
if (!sw) {
return 0;
@@ -999,8 +994,8 @@ static int audio_get_free (SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->samples)) {
+ dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -1015,9 +1010,10 @@ 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)
+static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
+ size_t samples)
{
- int n;
+ size_t n;
if (hw->enabled) {
SWVoiceCap *sc;
@@ -1028,17 +1024,17 @@ static void audio_capture_mix_and_clear (HWVoiceOut
*hw, int rpos, int samples)
n = samples;
while (n) {
- int till_end_of_hw = hw->samples - rpos2;
- int to_write = MIN (till_end_of_hw, n);
- int bytes = to_write << hw->info.shift;
- int written;
+ size_t till_end_of_hw = hw->samples - rpos2;
+ size_t to_write = MIN(till_end_of_hw, n);
+ size_t bytes = to_write << hw->info.shift;
+ size_t 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, mixed %d\n",
- bytes, written);
+ dolog("Could not mix %zu bytes into a capture "
+ "buffer, mixed %zu\n",
+ bytes, written);
break;
}
n -= to_write;
@@ -1047,9 +1043,9 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw,
int rpos, int samples)
}
}
- n = MIN (samples, hw->samples - rpos);
- mixeng_clear (hw->mix_buf + rpos, n);
- mixeng_clear (hw->mix_buf, samples - n);
+ n = 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)
@@ -1058,16 +1054,16 @@ static void audio_run_out (AudioState *s)
SWVoiceOut *sw;
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
- int played;
- int live, free, nb_live, cleanup_required, prev_rpos;
+ size_t played, live, prev_rpos, free;
+ int nb_live, cleanup_required;
live = audio_pcm_hw_get_live_out (hw, &nb_live);
if (!nb_live) {
live = 0;
}
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
continue;
}
@@ -1102,13 +1098,13 @@ static void audio_run_out (AudioState *s)
played = hw->pcm_ops->run_out (hw, live);
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
- dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
- hw->rpos, hw->samples, played);
+ dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
+ hw->rpos, hw->samples, played);
hw->rpos = 0;
}
#ifdef DEBUG_OUT
- dolog ("played=%d\n", played);
+ dolog("played=%zu\n", played);
#endif
if (played) {
@@ -1123,8 +1119,8 @@ static void audio_run_out (AudioState *s)
}
if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
- dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
- played, sw->total_hw_samples_mixed);
+ dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
+ played, sw->total_hw_samples_mixed);
played = sw->total_hw_samples_mixed;
}
@@ -1164,7 +1160,7 @@ static void audio_run_in (AudioState *s)
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
- int captured = 0, min;
+ size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
captured = hw->pcm_ops->run_in(hw);
@@ -1179,7 +1175,7 @@ static void audio_run_in (AudioState *s)
sw->total_hw_samples_acquired -= min;
if (sw->active) {
- int avail;
+ size_t avail;
avail = audio_get_avail (sw);
if (avail > 0) {
@@ -1195,15 +1191,15 @@ 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;
+ size_t live, rpos, captured;
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
rpos = hw->rpos;
while (live) {
- int left = hw->samples - rpos;
- int to_capture = MIN (live, left);
+ size_t left = hw->samples - rpos;
+ size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
@@ -1226,8 +1222,8 @@ static void audio_run_capture (AudioState *s)
}
if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
- dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
- captured, sw->total_hw_samples_mixed);
+ dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
+ captured, sw->total_hw_samples_mixed);
captured = sw->total_hw_samples_mixed;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 091fe84a34..d1be58b40a 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,9 +43,9 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
- int live;
- int decr;
- int rpos;
+ size_t live;
+ size_t decr;
+ size_t rpos;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -397,9 +397,9 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const
char *fn_name)
return 0;
}
-static int coreaudio_run_out (HWVoiceOut *hw, int live)
+static size_t coreaudio_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
if (coreaudio_lock (core, "coreaudio_run_out")) {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 11594c3095..67aa1d0ad5 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -454,19 +454,20 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int dsound_run_out (HWVoiceOut *hw, int live)
+static size_t dsound_run_out(HWVoiceOut *hw, size_t live)
{
int err;
HRESULT hr;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
- int len, hwshift;
+ size_t len;
+ int hwshift;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
DWORD wpos, ppos, old_pos;
LPVOID p1, p2;
- int bufsize;
+ size_t bufsize;
dsound *s = ds->s;
AudiodevDsoundOptions *dso = &s->dev->u.dsound;
@@ -640,13 +641,13 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static int dsound_run_in (HWVoiceIn *hw)
+static size_t dsound_run_in(HWVoiceIn *hw)
{
int err;
HRESULT hr;
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
- int live, len, dead;
+ size_t live, len, dead;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index cbb02d9e49..0fb2629cf2 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -41,10 +41,10 @@ typedef struct NoVoiceIn {
int64_t old_ticks;
} NoVoiceIn;
-static int no_run_out (HWVoiceOut *hw, int live)
+static size_t no_run_out(HWVoiceOut *hw, size_t live)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- int decr, samples;
+ size_t decr, samples;
int64_t now;
int64_t ticks;
int64_t bytes;
@@ -52,7 +52,7 @@ static int no_run_out (HWVoiceOut *hw, int live)
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - no->old_ticks;
bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- bytes = MIN(bytes, INT_MAX);
+ bytes = MIN(bytes, SIZE_MAX);
samples = bytes >> hw->info.shift;
no->old_ticks = now;
@@ -92,12 +92,12 @@ static void no_fini_in (HWVoiceIn *hw)
(void) hw;
}
-static int no_run_in (HWVoiceIn *hw)
+static size_t no_run_in(HWVoiceIn *hw)
{
NoVoiceIn *no = (NoVoiceIn *) hw;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int samples = 0;
+ size_t live = audio_pcm_hw_get_live_in(hw);
+ size_t dead = hw->samples - live;
+ size_t samples = 0;
if (dead) {
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
@@ -106,7 +106,7 @@ static int no_run_in (HWVoiceIn *hw)
muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
no->old_ticks = now;
- bytes = MIN (bytes, INT_MAX);
+ bytes = MIN (bytes, SIZE_MAX);
samples = bytes >> hw->info.shift;
samples = MIN (samples, dead);
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index b99edbec17..1696933688 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -411,13 +411,14 @@ static void oss_write_pending (OSSVoiceOut *oss)
}
}
-static int oss_run_out (HWVoiceOut *hw, int live)
+static size_t oss_run_out(HWVoiceOut *hw, size_t live)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- int err, decr;
+ int err;
+ size_t decr;
struct audio_buf_info abinfo;
struct count_info cntinfo;
- int bufsize;
+ size_t bufsize;
bufsize = hw->samples << hw->info.shift;
@@ -476,8 +477,8 @@ static void oss_fini_out (HWVoiceOut *hw)
if (oss->mmapped) {
err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
- oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
else {
@@ -543,8 +544,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
0
);
if (oss->pcm_buf == MAP_FAILED) {
- oss_logerr (errno, "Failed to map %d bytes of DAC\n",
- hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
+ hw->samples << hw->info.shift);
}
else {
int err;
@@ -568,8 +569,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
if (!oss->mmapped) {
err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
- oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
}
@@ -581,7 +582,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
1 << hw->info.shift);
if (!oss->pcm_buf) {
dolog (
- "Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+ "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
hw->samples,
1 << hw->info.shift
);
@@ -693,8 +694,8 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!oss->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
oss_anal_close (&fd);
return -1;
}
@@ -714,17 +715,17 @@ static void oss_fini_in (HWVoiceIn *hw)
oss->pcm_buf = NULL;
}
-static int oss_run_in (HWVoiceIn *hw)
+static size_t oss_run_in(HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
int hwshift = hw->info.shift;
int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
+ size_t live = audio_pcm_hw_get_live_in (hw);
+ size_t dead = hw->samples - live;
size_t read_samples = 0;
struct {
- int add;
- int len;
+ size_t add;
+ size_t len;
} bufs[2] = {
{ .add = hw->wpos, .len = 0 },
{ .add = 0, .len = 0 }
@@ -751,9 +752,9 @@ static int oss_run_in (HWVoiceIn *hw)
if (nread > 0) {
if (nread & hw->info.align) {
- dolog ("warning: Misaligned read %zd (requested %d), "
- "alignment %d\n", nread, bufs[i].add << hwshift,
- hw->info.align + 1);
+ dolog("warning: Misaligned read %zd (requested %zu), "
+ "alignment %d\n", nread, bufs[i].add << hwshift,
+ hw->info.align + 1);
}
read_samples += nread >> hwshift;
hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift);
@@ -766,9 +767,9 @@ static int oss_run_in (HWVoiceIn *hw)
case EAGAIN:
break;
default:
- oss_logerr (
+ oss_logerr(
errno,
- "Failed to read %d bytes of audio (to %p)\n",
+ "Failed to read %zu bytes of audio (to %p)\n",
bufs[i].len, p
);
break;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index efb72ced30..bfef9acaad 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -30,30 +30,30 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
- int done;
- int live;
- int decr;
- int rpos;
+ size_t done;
+ size_t live;
+ size_t decr;
+ size_t rpos;
pa_stream *stream;
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
- int samples;
+ size_t samples;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
- int done;
- int dead;
- int incr;
- int wpos;
+ size_t done;
+ size_t dead;
+ size_t incr;
+ size_t wpos;
pa_stream *stream;
void *pcm_buf;
struct audio_pt pt;
const void *read_data;
size_t read_index, read_length;
paaudio *g;
- int samples;
+ size_t samples;
} PAVoiceIn;
static void qpa_conn_fini(PAConnection *c);
@@ -219,7 +219,7 @@ static void *qpa_thread_out (void *arg)
}
for (;;) {
- int decr, to_mix, rpos;
+ size_t decr, to_mix, rpos;
for (;;) {
if (pa->done) {
@@ -244,7 +244,7 @@ static void *qpa_thread_out (void *arg)
while (to_mix) {
int error;
- int chunk = MIN (to_mix, hw->samples - rpos);
+ size_t chunk = MIN (to_mix, hw->samples - rpos);
struct st_sample *src = hw->mix_buf + rpos;
hw->clip (pa->pcm_buf, src, chunk);
@@ -273,9 +273,9 @@ static void *qpa_thread_out (void *arg)
return NULL;
}
-static int qpa_run_out (HWVoiceOut *hw, int live)
+static size_t qpa_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
PAVoiceOut *pa = (PAVoiceOut *) hw;
if (audio_pt_lock(&pa->pt, __func__)) {
@@ -306,7 +306,7 @@ static void *qpa_thread_in (void *arg)
}
for (;;) {
- int incr, to_grab, wpos;
+ size_t incr, to_grab, wpos;
for (;;) {
if (pa->done) {
@@ -331,7 +331,7 @@ static void *qpa_thread_in (void *arg)
while (to_grab) {
int error;
- int chunk = MIN (to_grab, hw->samples - wpos);
+ size_t chunk = MIN (to_grab, hw->samples - wpos);
void *buf = advance (pa->pcm_buf, wpos);
if (qpa_simple_read (pa, buf,
@@ -359,9 +359,9 @@ static void *qpa_thread_in (void *arg)
return NULL;
}
-static int qpa_run_in (HWVoiceIn *hw)
+static size_t qpa_run_in(HWVoiceIn *hw)
{
- int live, incr, dead;
+ size_t live, incr, dead;
PAVoiceIn *pa = (PAVoiceIn *) hw;
if (audio_pt_lock(&pa->pt, __func__)) {
@@ -582,8 +582,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as,
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
goto fail2;
}
@@ -650,8 +650,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
goto fail2;
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index c7fd487e0e..14b11f0335 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -41,8 +41,8 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
- int live;
- int decr;
+ size_t live;
+ size_t decr;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -184,22 +184,22 @@ static void sdl_callback (void *opaque, Uint8 *buf, int
len)
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
- int samples = len >> hw->info.shift;
- int to_mix, decr;
+ size_t samples = len >> hw->info.shift;
+ size_t to_mix, decr;
if (s->exit || !sdl->live) {
return;
}
- /* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */
+ /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
to_mix = MIN(samples, sdl->live);
decr = to_mix;
while (to_mix) {
- int chunk = MIN(to_mix, hw->samples - hw->rpos);
+ size_t chunk = MIN(to_mix, hw->samples - hw->rpos);
struct st_sample *src = hw->mix_buf + hw->rpos;
- /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
+ /* dolog ("in callback to_mix %zu, chunk %zu\n", to_mix, chunk); */
hw->clip(buf, src, chunk);
hw->rpos = (hw->rpos + chunk) % hw->samples;
to_mix -= chunk;
@@ -209,7 +209,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
sdl->live -= decr;
sdl->decr += decr;
- /* dolog ("done len=%d\n", len); */
+ /* dolog ("done len=%zu\n", len); */
/* SDL2 does not clear the remaining buffer for us, so do it on our own */
if (samples) {
@@ -217,9 +217,9 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
}
-static int sdl_run_out (HWVoiceOut *hw, int live)
+static size_t sdl_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDL_LockAudio();
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6e7f13cffa..d9db1a39e6 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -153,11 +153,11 @@ static void line_out_fini (HWVoiceOut *hw)
spice_server_remove_interface (&out->sin.base);
}
-static int line_out_run (HWVoiceOut *hw, int live)
+static size_t line_out_run (HWVoiceOut *hw, size_t live)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- int rpos, decr;
- int samples;
+ size_t rpos, decr;
+ size_t samples;
if (!live) {
return 0;
@@ -276,12 +276,12 @@ static void line_in_fini (HWVoiceIn *hw)
spice_server_remove_interface (&in->sin.base);
}
-static int line_in_run (HWVoiceIn *hw)
+static size_t line_in_run(HWVoiceIn *hw)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- int num_samples;
+ size_t num_samples;
int ready;
- int len[2];
+ size_t len[2];
uint64_t delta_samp;
const uint32_t *samples;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 17ab921cef..b6eeeb4e26 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -40,10 +40,10 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-static int wav_run_out (HWVoiceOut *hw, int live)
+static size_t wav_run_out(HWVoiceOut *hw, size_t live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int rpos, decr, samples;
+ size_t rpos, decr, samples;
uint8_t *dst;
struct st_sample *src;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
@@ -139,8 +139,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings
*as,
hw->samples = 1024;
wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!wav->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
return -1;
}
diff --git a/replay/replay-audio.c b/replay/replay-audio.c
index b113836de4..efe1628727 100644
--- a/replay/replay-audio.c
+++ b/replay/replay-audio.c
@@ -16,18 +16,18 @@
#include "sysemu/sysemu.h"
#include "audio/audio.h"
-void replay_audio_out(int *played)
+void replay_audio_out(size_t *played)
{
if (replay_mode == REPLAY_MODE_RECORD) {
g_assert(replay_mutex_locked());
replay_save_instructions();
replay_put_event(EVENT_AUDIO_OUT);
- replay_put_dword(*played);
+ replay_put_qword(*played);
} else if (replay_mode == REPLAY_MODE_PLAY) {
g_assert(replay_mutex_locked());
replay_account_executed_instructions();
if (replay_next_event_is(EVENT_AUDIO_OUT)) {
- *played = replay_get_dword();
+ *played = replay_get_qword();
replay_finish_event();
} else {
error_report("Missing audio out event in the replay log");
@@ -36,7 +36,7 @@ void replay_audio_out(int *played)
}
}
-void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
+void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t
size)
{
int pos;
uint64_t left, right;
@@ -44,8 +44,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos,
int size)
g_assert(replay_mutex_locked());
replay_save_instructions();
replay_put_event(EVENT_AUDIO_IN);
- replay_put_dword(*recorded);
- replay_put_dword(*wpos);
+ replay_put_qword(*recorded);
+ replay_put_qword(*wpos);
for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
; pos = (pos + 1) % size) {
audio_sample_to_uint64(samples, pos, &left, &right);
@@ -56,8 +56,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos,
int size)
g_assert(replay_mutex_locked());
replay_account_executed_instructions();
if (replay_next_event_is(EVENT_AUDIO_IN)) {
- *recorded = replay_get_dword();
- *wpos = replay_get_dword();
+ *recorded = replay_get_qword();
+ *wpos = replay_get_qword();
for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
; pos = (pos + 1) % size) {
left = replay_get_qword();
diff --git a/replay/replay.c b/replay/replay.c
index 8b172b2d1b..a1135a9943 100644
--- a/replay/replay.c
+++ b/replay/replay.c
@@ -22,7 +22,7 @@
/* Current version of the replay mechanism.
Increase it when file format changes. */
-#define REPLAY_VERSION 0xe02007
+#define REPLAY_VERSION 0xe02008
/* Size of replay log header */
#define HEADER_SIZE (sizeof(uint32_t) + sizeof(uint64_t))
--
2.22.0
- [Qemu-devel] [PATCH v3 06/14] audio: audiodev= parameters no longer optional when -audiodev present, (continued)
- [Qemu-devel] [PATCH v3 06/14] audio: audiodev= parameters no longer optional when -audiodev present, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 12/14] audio: remove read and write pcm_ops, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 08/14] paaudio: properly disconnect streams in fini_*, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 07/14] paaudio: do not move stream when sink/source name is specified, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 09/14] audio: remove audio_MIN, audio_MAX, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 14/14] audio: fix memory leak reported by ASAN, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 10/14] audio: do not run each backend in audio_run, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 05/14] paaudio: prepare for multiple audiodev, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 01/14] audio: reduce glob_audio_state usage, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 04/14] audio: add audiodev properties to frontends, Kővágó, Zoltán, 2019/08/04
- [Qemu-devel] [PATCH v3 13/14] audio: use size_t where makes sense,
Kővágó, Zoltán <=
- Re: [Qemu-devel] [PATCH v3 00/14] Multiple simultaneous audio backends, no-reply, 2019/08/04
- Re: [Qemu-devel] [PATCH v3 00/14] Multiple simultaneous audio backends, Gerd Hoffmann, 2019/08/13