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[Qemu-devel] [PATCH v2 34/49] audio: use size_t where makes sense
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From: |
Kővágó, Zoltán |
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Subject: |
[Qemu-devel] [PATCH v2 34/49] audio: use size_t where makes sense |
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Date: |
Fri, 21 Aug 2015 17:37:30 +0200 |
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
audio/alsaaudio.c | 8 +--
audio/audio.c | 163 ++++++++++++++++++++++++-------------------------
audio/audio.h | 4 +-
audio/audio_int.h | 22 +++----
audio/audio_template.h | 6 +-
audio/mixeng.h | 11 ++--
audio/ossaudio.c | 18 +++---
audio/paaudio.c | 8 +--
audio/rate_template.h | 2 +-
audio/sdlaudio.c | 4 +-
audio/wavaudio.c | 4 +-
11 files changed, 125 insertions(+), 125 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 057cad5..8199b71 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -748,8 +748,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as,
alsa->pcm_buf = audio_calloc (AUDIO_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;
}
@@ -850,8 +850,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
alsa->pcm_buf = audio_calloc (AUDIO_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;
}
diff --git a/audio/audio.c b/audio/audio.c
index 6d04dfe..013d980 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -499,10 +499,10 @@ 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) {
@@ -512,28 +512,28 @@ 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 (AUDIO_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(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu 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);
@@ -547,14 +547,14 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ dolog("live=%zd samples=%zu\n", live, hw->samples);
return 0;
}
@@ -567,17 +567,17 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live_in=%zu samples=%zu\n", live, hw->samples);
return 0;
}
@@ -591,9 +591,9 @@ 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;
}
@@ -602,11 +602,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int
size)
}
osamp = swlim;
- if (audio_bug (AUDIO_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;
@@ -627,10 +622,10 @@ 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) {
@@ -644,9 +639,9 @@ 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);
@@ -655,10 +650,10 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
@@ -669,10 +664,11 @@ 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;
+ int ret = 0;
+ size_t pos = 0, total = 0;
if (!sw) {
return size;
@@ -681,8 +677,8 @@ 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 (AUDIO_FUNC, live < 0 || live > hwsamples)){
- dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ if (audio_bug(AUDIO_FUNC, live > hwsamples)){
+ dolog("live=%zu samples=%zu\n", live, hwsamples);
return 0;
}
@@ -798,7 +794,7 @@ 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 */
@@ -813,7 +809,7 @@ 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 */
@@ -922,17 +918,17 @@ 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 (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -945,9 +941,9 @@ 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;
@@ -955,8 +951,8 @@ static int audio_get_free (SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -971,9 +967,10 @@ 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;
@@ -984,17 +981,17 @@ 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;
@@ -1003,9 +1000,9 @@ 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)
@@ -1014,16 +1011,16 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, hw->samples);
continue;
}
@@ -1056,14 +1053,14 @@ static void audio_run_out (AudioState *s)
prev_rpos = hw->rpos;
played = hw->pcm_ops->run_out (hw, live);
- if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
- dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
- hw->rpos, hw->samples, played);
+ if (audio_bug(AUDIO_FUNC, hw->rpos >= hw->samples)) {
+ dolog("rpos=%zu 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) {
@@ -1078,8 +1075,8 @@ static void audio_run_out (AudioState *s)
}
if (audio_bug (AUDIO_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;
}
@@ -1119,7 +1116,7 @@ static void audio_run_in (AudioState *s)
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
- int captured, min;
+ size_t captured, min;
captured = hw->pcm_ops->run_in (hw);
@@ -1131,7 +1128,7 @@ 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) {
@@ -1147,15 +1144,15 @@ 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;
@@ -1178,8 +1175,8 @@ static void audio_run_capture (AudioState *s)
}
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);
+ dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
+ captured, sw->total_hw_samples_mixed);
captured = sw->total_hw_samples_mixed;
}
@@ -1511,8 +1508,8 @@ CaptureVoiceOut *AUD_add_capture(
hw->mix_buf = audio_calloc (AUDIO_FUNC, hw->samples,
sizeof (struct st_sample));
if (!hw->mix_buf) {
- dolog ("Could not allocate capture mix buffer (%d samples)\n",
- hw->samples);
+ dolog("Could not allocate capture mix buffer (%zu samples)\n",
+ hw->samples);
goto err2;
}
@@ -1521,7 +1518,7 @@ CaptureVoiceOut *AUD_add_capture(
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",
+ "(%zu samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
goto err3;
}
diff --git a/audio/audio.h b/audio/audio.h
index d85138a..a48411a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -116,7 +116,7 @@ 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);
@@ -137,7 +137,7 @@ 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 59ccf10..a3bf79e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -59,12 +59,12 @@ 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;
@@ -80,13 +80,13 @@ 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;
@@ -101,7 +101,7 @@ 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;
@@ -118,7 +118,7 @@ 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;
@@ -207,10 +207,10 @@ extern struct audio_driver *drvtab[];
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);
@@ -223,7 +223,7 @@ 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 5a3dc90..5e28aea 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -79,8 +79,8 @@ static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
{
HWBUF = audio_calloc (AUDIO_FUNC, hw->samples, sizeof (struct st_sample));
if (!HWBUF) {
- dolog ("Could not allocate " NAME " buffer (%d samples)\n",
- hw->samples);
+ dolog("Could not allocate " NAME " buffer (%zu samples)\n",
+ hw->samples);
return -1;
}
@@ -265,7 +265,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
}
if (audio_bug (AUDIO_FUNC, hw->samples <= 0)) {
- dolog ("hw->samples=%d\n", hw->samples);
+ dolog("hw->samples=%zu\n", hw->samples);
goto err1;
}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 9de443b..013d894 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -24,6 +24,8 @@
#ifndef QEMU_MIXENG_H
#define QEMU_MIXENG_H
+#include <stddef.h>
+
#ifdef FLOAT_MIXENG
typedef float mixeng_real;
struct mixeng_volume { int mute; mixeng_real r; mixeng_real l; };
@@ -32,6 +34,7 @@ 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);
@@ -40,10 +43,10 @@ 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/ossaudio.c b/audio/ossaudio.c
index 30a872b..a4a9a5a 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -478,8 +478,8 @@ 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 {
@@ -545,8 +545,8 @@ 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;
@@ -570,8 +570,8 @@ 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);
}
}
}
@@ -585,7 +585,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
);
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
);
@@ -697,8 +697,8 @@ 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 (AUDIO_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;
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 1662c2c..98e5365 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -583,8 +583,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as,
pa->pcm_buf = audio_calloc (AUDIO_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;
}
@@ -644,8 +644,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
pa->pcm_buf = audio_calloc (AUDIO_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/rate_template.h b/audio/rate_template.h
index bd4b1c7..dc5e021 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/audio/sdlaudio.c b/audio/sdlaudio.c
index 470e0b9..26538c8 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -250,8 +250,8 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
- dolog ("sdl->live=%d hw->samples=%d\n",
- sdl->live, hw->samples);
+ dolog("sdl->live=%d hw->samples=%zu\n",
+ sdl->live, hw->samples);
return;
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index a12c934..edc78d5 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -136,8 +136,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings
*as,
hw->samples = 1024;
wav->pcm_buf = audio_calloc (AUDIO_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;
}
--
2.5.0
- [Qemu-devel] [PATCH v2 27/49] paaudio: do not create multiple connections to the same server, (continued)
- [Qemu-devel] [PATCH v2 27/49] paaudio: do not create multiple connections to the same server, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 28/49] paaudio: do not move stream when sink/source name is specified, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 31/49] audio: do not run each backend in audio_run, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 32/49] paaudio: fix playback glitches, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 33/49] audio: remove read and write pcm_ops, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 26/49] audio: audiodev= parameters no longer optional when -audiodev present, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 30/49] audio: remove audio_MIN, audio_MAX, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 35/49] audio: api for mixeng code free backends, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 37/49] coreaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 39/49] noaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 34/49] audio: use size_t where makes sense,
Kővágó, Zoltán <=
- [Qemu-devel] [PATCH v2 38/49] dsoundaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 40/49] ossaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 42/49] sdlaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 44/49] wavaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 45/49] audio: remove remains of the old backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 41/49] paaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 36/49] alsaaudio: port to the new audio backend api, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 46/49] audio: unify input and output mixeng buffer management, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 47/49] audio: remove hw->samples, buffer_size_in/out pcm_ops, Kővágó, Zoltán, 2015/08/21
- [Qemu-devel] [PATCH v2 48/49] audio: common rate control code for timer based outputs, Kővágó, Zoltán, 2015/08/21