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[Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
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From: |
Kővágó, Zoltán |
|
Subject: |
[Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX |
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Date: |
Sun, 23 Dec 2018 21:51:59 +0100 |
There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
Changes from v1:
* removed audio_MIN, audio_MAX macros
---
audio/alsaaudio.c | 6 +++---
audio/audio.c | 20 ++++++++++----------
audio/audio.h | 17 -----------------
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/alsaaudio.c b/audio/alsaaudio.c
index 6f75644538..d4bb540cb9 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -642,7 +642,7 @@ 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) {
@@ -705,7 +705,7 @@ 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);
@@ -923,7 +923,7 @@ 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 a7f79f3929..a32b86a813 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -537,7 +537,7 @@ 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;
@@ -557,14 +557,14 @@ 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);
@@ -618,7 +618,7 @@ 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;
@@ -666,7 +666,7 @@ 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;
}
}
@@ -729,7 +729,7 @@ 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);
@@ -741,7 +741,7 @@ 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;
}
@@ -1033,7 +1033,7 @@ 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;
@@ -1051,7 +1051,7 @@ 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);
}
@@ -1207,7 +1207,7 @@ 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/audio.h b/audio/audio.h
index ccfef9e10a..bcbe56d639 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -148,23 +148,6 @@ 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/coreaudio.c b/audio/coreaudio.c
index e00b8847d7..b6935359ee 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 a7d04b5033..6f074c6f94 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 f1eb048d80..299ba3d09c 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -51,11 +51,11 @@ 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;
}
@@ -110,9 +110,9 @@ 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;
}
@@ -123,7 +123,7 @@ 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 83ff1d2ebd..ae06e58d13 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 fa867a8065..6ed12851c0 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -234,7 +234,7 @@ 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__)) {
@@ -243,7 +243,7 @@ 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);
@@ -281,7 +281,7 @@ 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;
@@ -326,7 +326,7 @@ 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__)) {
@@ -335,7 +335,7 @@ 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,
@@ -374,7 +374,7 @@ 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 cf6ac19927..fc61038b77 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -288,10 +288,10 @@ static void sdl_callback (void *opaque, Uint8 *buf, int
len)
#endif
/* dolog ("in callback live=%d\n", 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); */
@@ -347,7 +347,7 @@ 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;
#if USE_SEMAPHORE
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 4f7873af5a..b9160991d2 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 214e30ccd9..723028a466 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -57,12 +57,12 @@ 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 03e3da4f31..1211541888 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -963,7 +963,7 @@ 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;
@@ -972,7 +972,7 @@ 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",
@@ -1018,7 +1018,7 @@ 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)
@@ -1039,7 +1039,7 @@ 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;
@@ -1048,7 +1048,7 @@ 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 0d01cd07c5..06350abf2c 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -194,7 +194,7 @@ 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);
@@ -209,7 +209,7 @@ 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 d25a120b0f..70d0947907 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -533,7 +533,7 @@ 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);
}
@@ -576,7 +576,7 @@ 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 8c63912a82..234a05633e 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -644,7 +644,7 @@ 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;
@@ -653,7 +653,7 @@ 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;
@@ -671,7 +671,7 @@ 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 dfb74cf0d3..2a2a24eacc 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -115,7 +115,7 @@ 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);
@@ -130,7 +130,7 @@ 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);
@@ -190,7 +190,7 @@ 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 e66103aedf..88106f88b4 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -233,10 +233,10 @@ 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) {
@@ -261,13 +261,13 @@ 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;
@@ -297,10 +297,10 @@ 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) {
@@ -325,7 +325,7 @@ 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 */
@@ -340,7 +340,7 @@ 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 8739cb376a..1a583fc3b9 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -183,7 +183,7 @@ 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;
@@ -197,7 +197,7 @@ 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;
@@ -226,7 +226,7 @@ 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;
@@ -240,7 +240,7 @@ 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 1457cd1ac6..94530c9265 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -102,7 +102,7 @@ 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 5a6a880f2e..78bf0753bb 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1166,7 +1166,7 @@ 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 c86fd91f98..13337b9b08 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -68,7 +68,7 @@ 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);
}
@@ -99,7 +99,7 @@ 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);
}
--
2.20.1
- [Qemu-devel] [PATCH v2 20/52] audio: audiodev= parameters no longer optional when -audiodev present, (continued)
- [Qemu-devel] [PATCH v2 20/52] audio: audiodev= parameters no longer optional when -audiodev present, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 18/52] audio: basic support for multi backend audio, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 24/52] audio: do not run each backend in audio_run, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 19/52] audio: add audiodev properties to frontends, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 25/52] paaudio: fix playback glitches, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 28/52] audio: api for mixeng code free backends, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense, Kővágó, Zoltán, 2018/12/23
- [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX,
Kővágó, Zoltán <=
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Philippe Mathieu-Daudé, 2018/12/23
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Zoltán Kővágó, 2018/12/23
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Philippe Mathieu-Daudé, 2018/12/24
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Kővágó Zoltán, 2018/12/24
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Philippe Mathieu-Daudé, 2018/12/25
- Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX, Kővágó Zoltán, 2018/12/27
[Qemu-devel] [PATCH v2 36/52] spiceaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
[Qemu-devel] [PATCH v2 32/52] noaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
[Qemu-devel] [PATCH v2 31/52] dsoundaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23
[Qemu-devel] [PATCH v2 33/52] ossaudio: port to the new audio backend api, Kővágó, Zoltán, 2018/12/23