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From: "Kővágó, Zoltán" <dirty.ice.hu@gmail.com>
To: qemu-devel@nongnu.org
Cc: "Marc-André Lureau" <marcandre.lureau@redhat.com>,
	"Michael Walle" <michael@walle.cc>,
	"Gerd Hoffmann" <kraxel@redhat.com>
Subject: [Qemu-devel] [PATCH v4 09/14] audio: remove audio_MIN, audio_MAX
Date: Mon, 19 Aug 2019 01:06:54 +0200
Message-ID: <303222477df6f7373217e0df768635fab5855745.1566168923.git.DirtY.iCE.hu@gmail.com> (raw)
In-Reply-To: <cover.1566168923.git.DirtY.iCE.hu@gmail.com>

There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/audio.h             | 17 -----------------
 audio/alsaaudio.c         |  6 +++---
 audio/audio.c             | 20 ++++++++++----------
 audio/coreaudio.c         |  2 +-
 audio/dsoundaudio.c       |  2 +-
 audio/noaudio.c           | 10 +++++-----
 audio/ossaudio.c          |  6 +++---
 audio/paaudio.c           | 12 ++++++------
 audio/sdlaudio.c          |  6 +++---
 audio/spiceaudio.c        | 10 +++++-----
 audio/wavaudio.c          |  4 ++--
 hw/audio/ac97.c           | 10 +++++-----
 hw/audio/adlib.c          |  4 ++--
 hw/audio/cs4231a.c        |  4 ++--
 hw/audio/es1370.c         |  6 +++---
 hw/audio/gus.c            |  6 +++---
 hw/audio/hda-codec.c      | 16 ++++++++--------
 hw/audio/milkymist-ac97.c |  8 ++++----
 hw/audio/pcspk.c          |  2 +-
 hw/audio/sb16.c           |  2 +-
 hw/audio/wm8750.c         |  4 ++--
 21 files changed, 70 insertions(+), 87 deletions(-)

diff --git a/audio/audio.h b/audio/audio.h
index 3257165b05..96e22887a0 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -147,23 +147,6 @@ static inline void *advance (void *p, int incr)
     return (d + incr);
 }
 
-#ifdef __GNUC__
-#define audio_MIN(a, b) ( __extension__ ({      \
-    __typeof (a) ta = a;                        \
-    __typeof (b) tb = b;                        \
-    ((ta)>(tb)?(tb):(ta));                      \
-}))
-
-#define audio_MAX(a, b) ( __extension__ ({      \
-    __typeof (a) ta = a;                        \
-    __typeof (b) tb = b;                        \
-    ((ta)<(tb)?(tb):(ta));                      \
-}))
-#else
-#define audio_MIN(a, b) ((a)>(b)?(b):(a))
-#define audio_MAX(a, b) ((a)<(b)?(b):(a))
-#endif
-
 int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
                       int freq, int bits, int nchannels);
 
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 3745c823ad..6b9e0f06af 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -634,7 +634,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
 
     while (alsa->pending) {
         int left_till_end_samples = hw->samples - alsa->wpos;
-        int len = audio_MIN (alsa->pending, left_till_end_samples);
+        int len = MIN (alsa->pending, left_till_end_samples);
         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 
         while (len) {
@@ -697,7 +697,7 @@ static int alsa_run_out (HWVoiceOut *hw, int live)
         return 0;
     }
 
-    decr = audio_MIN (live, avail);
+    decr = MIN (live, avail);
     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
     alsa->pending += decr;
     alsa_write_pending (alsa);
@@ -915,7 +915,7 @@ static int alsa_run_in (HWVoiceIn *hw)
         }
     }
 
-    decr = audio_MIN (dead, avail);
+    decr = MIN (dead, avail);
     if (!decr) {
         return 0;
     }
diff --git a/audio/audio.c b/audio/audio.c
index 7267cab9d7..100c3ea177 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -533,7 +533,7 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
 
     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
         if (sw->active) {
-            m = audio_MIN (m, sw->total_hw_samples_acquired);
+            m = MIN (m, sw->total_hw_samples_acquired);
         }
     }
     return m;
@@ -553,14 +553,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
                            int live, int pending)
 {
     int left = hw->samples - pending;
-    int len = audio_MIN (left, live);
+    int len = MIN (left, live);
     int 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 = audio_MIN (len, samples_till_end_of_buf);
+        int samples_to_clip = MIN (len, samples_till_end_of_buf);
 
         hw->clip (dst, src, samples_to_clip);
 
@@ -614,7 +614,7 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
     }
 
     swlim = (live * sw->ratio) >> 32;
-    swlim = audio_MIN (swlim, samples);
+    swlim = MIN (swlim, samples);
 
     while (swlim) {
         src = hw->conv_buf + rpos;
@@ -662,7 +662,7 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
 
     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
         if (sw->active || !sw->empty) {
-            m = audio_MIN (m, sw->total_hw_samples_mixed);
+            m = MIN (m, sw->total_hw_samples_mixed);
             nb_live += 1;
         }
     }
@@ -725,7 +725,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
 
     dead = hwsamples - live;
     swlim = ((int64_t) dead << 32) / sw->ratio;
-    swlim = audio_MIN (swlim, samples);
+    swlim = MIN (swlim, samples);
     if (swlim) {
         sw->conv (sw->buf, buf, swlim);
 
@@ -737,7 +737,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
     while (swlim) {
         dead = hwsamples - live;
         left = hwsamples - wpos;
-        blck = audio_MIN (dead, left);
+        blck = MIN (dead, left);
         if (!blck) {
             break;
         }
@@ -1029,7 +1029,7 @@ 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 = audio_MIN (till_end_of_hw, n);
+                int to_write = MIN (till_end_of_hw, n);
                 int bytes = to_write << hw->info.shift;
                 int written;
 
@@ -1047,7 +1047,7 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
         }
     }
 
-    n = audio_MIN (samples, hw->samples - rpos);
+    n = MIN (samples, hw->samples - rpos);
     mixeng_clear (hw->mix_buf + rpos, n);
     mixeng_clear (hw->mix_buf, samples - n);
 }
@@ -1203,7 +1203,7 @@ static void audio_run_capture (AudioState *s)
         rpos = hw->rpos;
         while (live) {
             int left = hw->samples - rpos;
-            int to_capture = audio_MIN (live, left);
+            int to_capture = MIN (live, left);
             struct st_sample *src;
             struct capture_callback *cb;
 
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 4bec6c8c5c..f0ab4014a8 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -413,7 +413,7 @@ static int coreaudio_run_out (HWVoiceOut *hw, int live)
                 core->live);
     }
 
-    decr = audio_MIN (core->decr, live);
+    decr = MIN (core->decr, live);
     core->decr -= decr;
 
     core->live = live - decr;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 5da4c864c3..07260f881e 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -707,7 +707,7 @@ static int dsound_run_in (HWVoiceIn *hw)
     if (!len) {
         return 0;
     }
-    len = audio_MIN (len, dead);
+    len = MIN (len, dead);
 
     err = dsound_lock_in (
         dscb,
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 9b195dc52c..14a0e4ab29 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -52,11 +52,11 @@ 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 = audio_MIN(bytes, INT_MAX);
+    bytes = MIN(bytes, INT_MAX);
     samples = bytes >> hw->info.shift;
 
     no->old_ticks = now;
-    decr = audio_MIN (live, samples);
+    decr = MIN (live, samples);
     hw->rpos = (hw->rpos + decr) % hw->samples;
     return decr;
 }
@@ -111,9 +111,9 @@ static int no_run_in (HWVoiceIn *hw)
             muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
 
         no->old_ticks = now;
-        bytes = audio_MIN (bytes, INT_MAX);
+        bytes = MIN (bytes, INT_MAX);
         samples = bytes >> hw->info.shift;
-        samples = audio_MIN (samples, dead);
+        samples = MIN (samples, dead);
     }
     return samples;
 }
@@ -124,7 +124,7 @@ static int no_read (SWVoiceIn *sw, void *buf, int size)
      * useless resampling/mixing */
     int samples = size >> sw->info.shift;
     int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
-    int to_clear = audio_MIN (samples, total);
+    int to_clear = MIN (samples, total);
     sw->total_hw_samples_acquired += total;
     audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
     return to_clear << sw->info.shift;
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index c0af065b6f..29139ef1f5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -388,7 +388,7 @@ static void oss_write_pending (OSSVoiceOut *oss)
         int samples_written;
         ssize_t bytes_written;
         int samples_till_end = hw->samples - oss->wpos;
-        int samples_to_write = audio_MIN (oss->pending, samples_till_end);
+        int samples_to_write = MIN (oss->pending, samples_till_end);
         int bytes_to_write = samples_to_write << hw->info.shift;
         void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift);
 
@@ -437,7 +437,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
 
         pos = hw->rpos << hw->info.shift;
         bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize);
-        decr = audio_MIN (bytes >> hw->info.shift, live);
+        decr = MIN (bytes >> hw->info.shift, live);
     }
     else {
         err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
@@ -456,7 +456,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
             return 0;
         }
 
-        decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+        decr = MIN (abinfo.bytes >> hw->info.shift, live);
         if (!decr) {
             return 0;
         }
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 1d68173636..f3864e1d50 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -235,7 +235,7 @@ static void *qpa_thread_out (void *arg)
             }
         }
 
-        decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
+        decr = to_mix = MIN(pa->live, pa->samples >> 5);
         rpos = pa->rpos;
 
         if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -244,7 +244,7 @@ static void *qpa_thread_out (void *arg)
 
         while (to_mix) {
             int error;
-            int chunk = audio_MIN (to_mix, hw->samples - rpos);
+            int chunk = MIN (to_mix, hw->samples - rpos);
             struct st_sample *src = hw->mix_buf + rpos;
 
             hw->clip (pa->pcm_buf, src, chunk);
@@ -282,7 +282,7 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
         return 0;
     }
 
-    decr = audio_MIN (live, pa->decr);
+    decr = MIN (live, pa->decr);
     pa->decr -= decr;
     pa->live = live - decr;
     hw->rpos = pa->rpos;
@@ -327,7 +327,7 @@ static void *qpa_thread_in (void *arg)
             }
         }
 
-        incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
+        incr = to_grab = MIN(pa->dead, pa->samples >> 5);
         wpos = pa->wpos;
 
         if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -336,7 +336,7 @@ static void *qpa_thread_in (void *arg)
 
         while (to_grab) {
             int error;
-            int chunk = audio_MIN (to_grab, hw->samples - wpos);
+            int chunk = MIN (to_grab, hw->samples - wpos);
             void *buf = advance (pa->pcm_buf, wpos);
 
             if (qpa_simple_read (pa, buf,
@@ -375,7 +375,7 @@ static int qpa_run_in (HWVoiceIn *hw)
 
     live = audio_pcm_hw_get_live_in (hw);
     dead = hw->samples - live;
-    incr = audio_MIN (dead, pa->incr);
+    incr = MIN (dead, pa->incr);
     pa->incr -= incr;
     pa->dead = dead - incr;
     hw->wpos = pa->wpos;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index e7179ff1d4..42f7614124 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -193,10 +193,10 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
     /* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */
 
-    to_mix = audio_MIN(samples, sdl->live);
+    to_mix = MIN(samples, sdl->live);
     decr = to_mix;
     while (to_mix) {
-        int chunk = audio_MIN(to_mix, hw->samples - hw->rpos);
+        int 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); */
@@ -236,7 +236,7 @@ static int sdl_run_out (HWVoiceOut *hw, int live)
                 sdl->live);
     }
 
-    decr = audio_MIN (sdl->decr, live);
+    decr = MIN (sdl->decr, live);
     sdl->decr -= decr;
 
     sdl->live = live;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index ec1c8fe936..5672e12dc3 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -163,20 +163,20 @@ static int line_out_run (HWVoiceOut *hw, int live)
     }
 
     decr = rate_get_samples (&hw->info, &out->rate);
-    decr = audio_MIN (live, decr);
+    decr = MIN (live, decr);
 
     samples = decr;
     rpos = hw->rpos;
     while (samples) {
         int left_till_end_samples = hw->samples - rpos;
-        int len = audio_MIN (samples, left_till_end_samples);
+        int len = MIN (samples, left_till_end_samples);
 
         if (!out->frame) {
             spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize);
             out->fpos = out->frame;
         }
         if (out->frame) {
-            len = audio_MIN (len, out->fsize);
+            len = MIN (len, out->fsize);
             hw->clip (out->fpos, hw->mix_buf + rpos, len);
             out->fsize -= len;
             out->fpos  += len;
@@ -294,7 +294,7 @@ static int line_in_run (HWVoiceIn *hw)
     }
 
     delta_samp = rate_get_samples (&hw->info, &in->rate);
-    num_samples = audio_MIN (num_samples, delta_samp);
+    num_samples = MIN (num_samples, delta_samp);
 
     ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples);
     samples = in->samples;
@@ -304,7 +304,7 @@ static int line_in_run (HWVoiceIn *hw)
         ready = LINE_IN_SAMPLES;
     }
 
-    num_samples = audio_MIN (ready, num_samples);
+    num_samples = MIN (ready, num_samples);
 
     if (hw->wpos + num_samples > hw->samples) {
         len[0] = hw->samples - hw->wpos;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 803b6cb1f3..bbf3f3b346 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -59,12 +59,12 @@ static int wav_run_out (HWVoiceOut *hw, int live)
     }
 
     wav->old_ticks = now;
-    decr = audio_MIN (live, samples);
+    decr = MIN (live, samples);
     samples = decr;
     rpos = hw->rpos;
     while (samples) {
         int left_till_end_samples = hw->samples - rpos;
-        int convert_samples = audio_MIN (samples, left_till_end_samples);
+        int convert_samples = MIN (samples, left_till_end_samples);
 
         src = hw->mix_buf + rpos;
         dst = advance (wav->pcm_buf, rpos << hw->info.shift);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index b4c0b690d7..a136b97f68 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -965,7 +965,7 @@ static int write_audio (AC97LinkState *s, AC97BusMasterRegs *r,
     uint32_t temp = r->picb << 1;
     uint32_t written = 0;
     int to_copy = 0;
-    temp = audio_MIN (temp, max);
+    temp = MIN (temp, max);
 
     if (!temp) {
         *stop = 1;
@@ -974,7 +974,7 @@ static int write_audio (AC97LinkState *s, AC97BusMasterRegs *r,
 
     while (temp) {
         int copied;
-        to_copy = audio_MIN (temp, sizeof (tmpbuf));
+        to_copy = MIN (temp, sizeof (tmpbuf));
         pci_dma_read (&s->dev, addr, tmpbuf, to_copy);
         copied = AUD_write (s->voice_po, tmpbuf, to_copy);
         dolog ("write_audio max=%x to_copy=%x copied=%x\n",
@@ -1020,7 +1020,7 @@ static void write_bup (AC97LinkState *s, int elapsed)
     }
 
     while (elapsed) {
-        int temp = audio_MIN (elapsed, sizeof (s->silence));
+        int temp = MIN (elapsed, sizeof (s->silence));
         while (temp) {
             int copied = AUD_write (s->voice_po, s->silence, temp);
             if (!copied)
@@ -1041,7 +1041,7 @@ static int read_audio (AC97LinkState *s, AC97BusMasterRegs *r,
     int to_copy = 0;
     SWVoiceIn *voice = (r - s->bm_regs) == MC_INDEX ? s->voice_mc : s->voice_pi;
 
-    temp = audio_MIN (temp, max);
+    temp = MIN (temp, max);
 
     if (!temp) {
         *stop = 1;
@@ -1050,7 +1050,7 @@ static int read_audio (AC97LinkState *s, AC97BusMasterRegs *r,
 
     while (temp) {
         int acquired;
-        to_copy = audio_MIN (temp, sizeof (tmpbuf));
+        to_copy = MIN (temp, sizeof (tmpbuf));
         acquired = AUD_read (voice, tmpbuf, to_copy);
         if (!acquired) {
             *stop = 1;
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 64a1060b80..cb4178d861 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -195,7 +195,7 @@ static void adlib_callback (void *opaque, int free)
         return;
     }
 
-    to_play = audio_MIN (s->left, samples);
+    to_play = MIN (s->left, samples);
     while (to_play) {
         written = write_audio (s, to_play);
 
@@ -210,7 +210,7 @@ static void adlib_callback (void *opaque, int free)
         }
     }
 
-    samples = audio_MIN (samples, s->samples - s->pos);
+    samples = MIN (samples, s->samples - s->pos);
     if (!samples) {
         return;
     }
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 5b7baeae25..c7b8067489 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -536,7 +536,7 @@ static int cs_write_audio (CSState *s, int nchan, int dma_pos,
         int copied;
         size_t to_copy;
 
-        to_copy = audio_MIN (temp, left);
+        to_copy = MIN (temp, left);
         if (to_copy > sizeof (tmpbuf)) {
             to_copy = sizeof (tmpbuf);
         }
@@ -579,7 +579,7 @@ static int cs_dma_read (void *opaque, int nchan, int dma_pos, int dma_len)
         till = (s->dregs[Playback_Lower_Base_Count]
             | (s->dregs[Playback_Upper_Base_Count] << 8)) << s->shift;
         till -= s->transferred;
-        copy = audio_MIN (till, copy);
+        copy = MIN (till, copy);
     }
 
     if ((copy <= 0) || (dma_len <= 0)) {
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 88f886491b..f9e9f2a3b3 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -645,7 +645,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
     int size = d->frame_cnt & 0xffff;
     int left = ((size - cnt + 1) << 2) + d->leftover;
     int transferred = 0;
-    int temp = audio_MIN (max, audio_MIN (left, csc_bytes));
+    int temp = MIN (max, MIN (left, csc_bytes));
     int index = d - &s->chan[0];
 
     addr += (cnt << 2) + d->leftover;
@@ -654,7 +654,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
         while (temp) {
             int acquired, to_copy;
 
-            to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
+            to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
             acquired = AUD_read (s->adc_voice, tmpbuf, to_copy);
             if (!acquired)
                 break;
@@ -672,7 +672,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
         while (temp) {
             int copied, to_copy;
 
-            to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
+            to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
             pci_dma_read (&s->dev, addr, tmpbuf, to_copy);
             copied = AUD_write (voice, tmpbuf, to_copy);
             if (!copied)
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index effec31d35..2b6b7c4e3f 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -119,7 +119,7 @@ static void GUS_callback (void *opaque, int free)
     GUSState *s = opaque;
 
     samples = free >> s->shift;
-    to_play = audio_MIN (samples, s->left);
+    to_play = MIN (samples, s->left);
 
     while (to_play) {
         int written = write_audio (s, to_play);
@@ -134,7 +134,7 @@ static void GUS_callback (void *opaque, int free)
         net += written;
     }
 
-    samples = audio_MIN (samples, s->samples);
+    samples = MIN (samples, s->samples);
     if (samples) {
         gus_mixvoices (&s->emu, s->freq, samples, s->mixbuf);
 
@@ -194,7 +194,7 @@ static int GUS_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
     ldebug ("read DMA %#x %d\n", dma_pos, dma_len);
     mode = k->has_autoinitialization(s->isa_dma, s->emu.gusdma);
     while (left) {
-        int to_copy = audio_MIN ((size_t) left, sizeof (tmpbuf));
+        int to_copy = MIN ((size_t) left, sizeof (tmpbuf));
         int copied;
 
         ldebug ("left=%d to_copy=%d pos=%d\n", left, to_copy, pos);
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 68fe1de987..f17e8d8dce 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -235,10 +235,10 @@ static void hda_audio_input_timer(void *opaque)
         goto out_timer;
     }
 
-    int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
+    int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos);
     while (to_transfer) {
         uint32_t start = (rpos & B_MASK);
-        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = MIN(B_SIZE - start, to_transfer);
         int rc = hda_codec_xfer(
                 &st->state->hda, st->stream, false, st->buf + start, chunk);
         if (!rc) {
@@ -263,13 +263,13 @@ static void hda_audio_input_cb(void *opaque, int avail)
     int64_t wpos = st->wpos;
     int64_t rpos = st->rpos;
 
-    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
+    int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
 
     hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
 
     while (to_transfer) {
         uint32_t start = (uint32_t) (wpos & B_MASK);
-        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
         uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
         wpos += read;
         to_transfer -= read;
@@ -299,10 +299,10 @@ static void hda_audio_output_timer(void *opaque)
         goto out_timer;
     }
 
-    int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
+    int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
     while (to_transfer) {
         uint32_t start = (wpos & B_MASK);
-        uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = MIN(B_SIZE - start, to_transfer);
         int rc = hda_codec_xfer(
                 &st->state->hda, st->stream, true, st->buf + start, chunk);
         if (!rc) {
@@ -327,7 +327,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
     int64_t wpos = st->wpos;
     int64_t rpos = st->rpos;
 
-    int64_t to_transfer = audio_MIN(wpos - rpos, avail);
+    int64_t to_transfer = MIN(wpos - rpos, avail);
 
     if (wpos - rpos == B_SIZE) {
         /* drop buffer, reset timer adjust */
@@ -342,7 +342,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
 
     while (to_transfer) {
         uint32_t start = (uint32_t) (rpos & B_MASK);
-        uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+        uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
         uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
         rpos += written;
         to_transfer -= written;
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 853b4ed874..6d409eff1b 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -185,7 +185,7 @@ static void ac97_in_cb(void *opaque, int avail_b)
     MilkymistAC97State *s = opaque;
     uint8_t buf[4096];
     uint32_t remaining = s->regs[R_U_REMAINING];
-    int temp = audio_MIN(remaining, avail_b);
+    int temp = MIN(remaining, avail_b);
     uint32_t addr = s->regs[R_U_ADDR];
     int transferred = 0;
 
@@ -199,7 +199,7 @@ static void ac97_in_cb(void *opaque, int avail_b)
     while (temp) {
         int acquired, to_copy;
 
-        to_copy = audio_MIN(temp, sizeof(buf));
+        to_copy = MIN(temp, sizeof(buf));
         acquired = AUD_read(s->voice_in, buf, to_copy);
         if (!acquired) {
             break;
@@ -228,7 +228,7 @@ static void ac97_out_cb(void *opaque, int free_b)
     MilkymistAC97State *s = opaque;
     uint8_t buf[4096];
     uint32_t remaining = s->regs[R_D_REMAINING];
-    int temp = audio_MIN(remaining, free_b);
+    int temp = MIN(remaining, free_b);
     uint32_t addr = s->regs[R_D_ADDR];
     int transferred = 0;
 
@@ -242,7 +242,7 @@ static void ac97_out_cb(void *opaque, int free_b)
     while (temp) {
         int copied, to_copy;
 
-        to_copy = audio_MIN(temp, sizeof(buf));
+        to_copy = MIN(temp, sizeof(buf));
         cpu_physical_memory_read(addr, buf, to_copy);
         copied = AUD_write(s->voice_out, buf, to_copy);
         if (!copied) {
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 307c664725..d773eb80de 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -103,7 +103,7 @@ static void pcspk_callback(void *opaque, int free)
     }
 
     while (free > 0) {
-        n = audio_MIN(s->samples - s->play_pos, (unsigned int)free);
+        n = MIN(s->samples - s->play_pos, (unsigned int)free);
         n = AUD_write(s->voice, &s->sample_buf[s->play_pos], n);
         if (!n)
             break;
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index 7e50ab8a84..a354f94acb 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1169,7 +1169,7 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
         int copied;
         size_t to_copy;
 
-        to_copy = audio_MIN (temp, left);
+        to_copy = MIN (temp, left);
         if (to_copy > sizeof (tmpbuf)) {
             to_copy = sizeof (tmpbuf);
         }
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index a8725abf6a..601ed04aff 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -70,7 +70,7 @@ static inline void wm8750_in_load(WM8750State *s)
 {
     if (s->idx_in + s->req_in <= sizeof(s->data_in))
         return;
-    s->idx_in = audio_MAX(0, (int) sizeof(s->data_in) - s->req_in);
+    s->idx_in = MAX(0, (int) sizeof(s->data_in) - s->req_in);
     AUD_read(*s->in[0], s->data_in + s->idx_in,
              sizeof(s->data_in) - s->idx_in);
 }
@@ -101,7 +101,7 @@ static void wm8750_audio_out_cb(void *opaque, int free_b)
         wm8750_out_flush(s);
     } else
         s->req_out = free_b - s->idx_out;
- 
+
     s->data_req(s->opaque, s->req_out >> 2, s->req_in >> 2);
 }
 
-- 
2.22.0



  parent reply index

Thread overview: 19+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2019-08-18 23:06 [Qemu-devel] [PATCH v4 00/14] Multiple simultaneous audio backends Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 01/14] audio: reduce glob_audio_state usage Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 02/14] audio: basic support for multi backend audio Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 03/14] audio: add audiodev property to vnc and wav_capture Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 04/14] audio: add audiodev properties to frontends Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 05/14] paaudio: prepare for multiple audiodev Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 06/14] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 07/14] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 08/14] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
2019-08-18 23:06 ` Kővágó, Zoltán [this message]
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 10/14] audio: do not run each backend in audio_run Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 11/14] paaudio: fix playback glitches Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 12/14] audio: remove read and write pcm_ops Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 13/14] audio: use size_t where makes sense Kővágó, Zoltán
2019-08-18 23:06 ` [Qemu-devel] [PATCH v4 14/14] audio: fix memory leak reported by ASAN Kővágó, Zoltán
2019-08-18 23:25   ` Philippe Mathieu-Daudé
2019-08-19  2:05     ` Zoltán Kővágó
2019-08-19 11:21       ` Philippe Mathieu-Daudé
2019-08-18 23:35 ` [Qemu-devel] [PATCH v4 00/14] Multiple simultaneous audio backends no-reply

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