qemu-devel
[Top][All Lists]
Advanced

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

[Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense


From: Kővágó, Zoltán
Subject: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
Date: Sun, 23 Dec 2018 21:52:03 +0100

Signed-off-by: Kővágó, Zoltán <address@hidden>
---
 audio/alsaaudio.c       |   8 +-
 audio/audio.c           | 162 ++++++++++++++++++++--------------------
 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        |   3 +-
 audio/wavaudio.c        |   4 +-
 include/sysemu/replay.h |   4 +-
 replay/replay-audio.c   |  16 ++--
 13 files changed, 133 insertions(+), 135 deletions(-)

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 19de7d01cb..69e7a3868c 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -747,8 +747,8 @@ 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;
     }
@@ -849,8 +849,8 @@ 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;
     }
diff --git a/audio/audio.c b/audio/audio.c
index 1ea80ba6a7..27a8a31a64 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -530,10 +530,10 @@ 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) {
@@ -543,28 +543,28 @@ 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);
 
@@ -578,14 +578,14 @@ 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;
     }
 
@@ -598,17 +598,17 @@ 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;
     }
 
@@ -622,9 +622,9 @@ 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;
         }
 
@@ -633,11 +633,6 @@ 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;
@@ -658,10 +653,10 @@ 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) {
@@ -675,9 +670,9 @@ 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);
@@ -686,10 +681,10 @@ 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;
@@ -700,10 +695,10 @@ 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;
@@ -712,8 +707,8 @@ 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;
     }
 
@@ -767,7 +762,7 @@ 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,
@@ -846,7 +841,7 @@ 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 */
@@ -861,7 +856,7 @@ 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 */
@@ -970,17 +965,17 @@ 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;
     }
 
@@ -993,9 +988,9 @@ 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;
@@ -1003,8 +998,8 @@ 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;
     }
 
@@ -1019,9 +1014,10 @@ 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;
@@ -1032,17 +1028,17 @@ 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;
@@ -1051,9 +1047,9 @@ 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)
@@ -1062,16 +1058,16 @@ 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;
         }
 
@@ -1106,13 +1102,13 @@ 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) {
@@ -1127,8 +1123,8 @@ 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;
             }
 
@@ -1168,7 +1164,7 @@ 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);
@@ -1183,7 +1179,7 @@ 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) {
@@ -1199,15 +1195,15 @@ 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;
 
@@ -1230,8 +1226,8 @@ 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;
             }
 
@@ -1570,8 +1566,8 @@ CaptureVoiceOut *AUD_add_capture(
         hw->mix_buf = audio_calloc(__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;
         }
 
@@ -1580,7 +1576,7 @@ CaptureVoiceOut *AUD_add_capture(
         cap->buf = audio_calloc(__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 bcbe56d639..bfb12e7531 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -114,7 +114,7 @@ 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);
@@ -135,7 +135,7 @@ 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 d269c38465..330c465d0b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -60,12 +60,12 @@ 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;
@@ -81,13 +81,13 @@ 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;
@@ -102,7 +102,7 @@ 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;
@@ -119,7 +119,7 @@ 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 @@ 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);
@@ -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 ce1e5d6559..fecbf1a046 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(__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(__func__, hw->samples <= 0)) {
-        dolog ("hw->samples=%d\n", hw->samples);
+        dolog("hw->samples=%zd\n", hw->samples);
         goto err1;
     }
 
diff --git a/audio/mixeng.h b/audio/mixeng.h
index b53a5ef99a..2c09ed41e7 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -25,6 +25,8 @@
 #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; };
@@ -33,6 +35,7 @@ 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 +44,10 @@ 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 70909e5696..05c7d6f85c 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -476,8 +476,8 @@ 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 +543,8 @@ 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 +568,8 @@ 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 +581,7 @@ 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 +693,8 @@ 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;
     }
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6a1919e93b..251b087a74 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -584,8 +584,8 @@ 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;
     }
 
@@ -645,8 +645,8 @@ 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/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/audio/sdlaudio.c b/audio/sdlaudio.c
index ab9166d054..92da4804c6 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -273,8 +273,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
         }
 
         if (audio_bug(__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 dda6993fb9..58300663ff 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -137,8 +137,8 @@ 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/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/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();
-- 
2.20.1




reply via email to

[Prev in Thread] Current Thread [Next in Thread]