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[Qemu-devel] [PULL 09/15] audio: remove audio_MIN, audio_MAX
|
From: |
Gerd Hoffmann |
|
Subject: |
[Qemu-devel] [PULL 09/15] audio: remove audio_MIN, audio_MAX |
|
Date: |
Tue, 13 Aug 2019 13:18:03 +0200 |
From: Kővágó, Zoltán <address@hidden>
There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.
Signed-off-by: Kővágó, Zoltán <address@hidden>
Reviewed-by: Marc-André Lureau <address@hidden>
Message-id: address@hidden
Signed-off-by: Gerd Hoffmann <address@hidden>
---
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 c0722a5cda94..4a9575851625 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -146,23 +146,6 @@ 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 3745c823ad37..6b9e0f06af47 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 d1319581949d..cb0222ab4a87 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 4bec6c8c5c13..f0ab4014a85d 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 5da4c864c3f2..07260f881eaa 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 9b195dc52ca3..14a0e4ab29f4 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 c0af065b6ff2..29139ef1f5cd 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 1d68173636ed..f3864e1d5038 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 e7179ff1d410..42f7614124c6 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 0ead5ae43ade..6f4a0558f86d 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -164,20 +164,20 @@ 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;
@@ -295,7 +295,7 @@ 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;
@@ -305,7 +305,7 @@ 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 803b6cb1f3d0..bbf3f3b3462f 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 0d8e52423364..060bafdac316 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -964,7 +964,7 @@ 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;
@@ -973,7 +973,7 @@ 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",
@@ -1019,7 +1019,7 @@ 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)
@@ -1040,7 +1040,7 @@ 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;
@@ -1049,7 +1049,7 @@ 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 df2e781788ab..1b32c4ff7f49 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 e3ea830b4707..ca3af8a9878c 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -535,7 +535,7 @@ 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);
}
@@ -578,7 +578,7 @@ 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 7589671d207b..50b144ded068 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 566864bc9e59..325efd8df7d7 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -117,7 +117,7 @@ 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);
@@ -132,7 +132,7 @@ 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);
@@ -192,7 +192,7 @@ 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 967a10f1893b..c8f513d3ff87 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -234,10 +234,10 @@ 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) {
@@ -262,13 +262,13 @@ 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;
@@ -298,10 +298,10 @@ 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) {
@@ -326,7 +326,7 @@ 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 */
@@ -341,7 +341,7 @@ 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 483522932635..929b856587e3 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -184,7 +184,7 @@ 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;
@@ -198,7 +198,7 @@ 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;
@@ -227,7 +227,7 @@ 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;
@@ -241,7 +241,7 @@ 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 01127304c239..6bb1455c1b1b 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 6b604979cf56..5182eba8eb06 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1168,7 +1168,7 @@ 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 dfb4156ff49f..ab04bfa2c397 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -69,7 +69,7 @@ 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);
}
@@ -100,7 +100,7 @@ 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);
}
--
2.18.1
- [Qemu-devel] [PULL 00/15] Audio 20190813 patches, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 03/15] audio: add audiodev property to vnc and wav_capture, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 04/15] audio: add audiodev properties to frontends, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 07/15] paaudio: do not move stream when sink/source name is specified, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 09/15] audio: remove audio_MIN, audio_MAX,
Gerd Hoffmann <=
- [Qemu-devel] [PULL 14/15] audio: fix memory leak reported by ASAN, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 05/15] paaudio: prepare for multiple audiodev, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 11/15] paaudio: fix playback glitches, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 02/15] audio: basic support for multi backend audio, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 08/15] paaudio: properly disconnect streams in fini_*, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 13/15] audio: use size_t where makes sense, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 06/15] audio: audiodev= parameters no longer optional when -audiodev present, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 15/15] audio: Add missing fall through comments, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 12/15] audio: remove read and write pcm_ops, Gerd Hoffmann, 2019/08/13
- [Qemu-devel] [PULL 10/15] audio: do not run each backend in audio_run, Gerd Hoffmann, 2019/08/13