Re: [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio

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Re: [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX

From:	Marc-André Lureau
Subject:	Re: [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX
Date:	Wed, 10 Jul 2019 23:58:26 +0400
On Tue, Jul 9, 2019 at 11:08 PM Kővágó, Zoltán <address@hidden> wrote:
>
> 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>


> ---
>  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 c0722a5cda..4a95758516 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 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 82dd0e3e13..6bf30ac9b3 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 490bcd770e..9d46f11b0a 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 0ead5ae43a..6f4a0558f8 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 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 0d8e524233..060bafdac3 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 df2e781788..1b32c4ff7f 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 e3ea830b47..ca3af8a987 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 7589671d20..50b144ded0 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 566864bc9e..325efd8df7 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 967a10f189..c8f513d3ff 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 4835229326..929b856587 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 01127304c2..6bb1455c1b 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 6b604979cf..5182eba8eb 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 dfb4156ff4..ab04bfa2c3 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);
>  }
>
> --
> 2.22.0
>
>


--
Marc-André Lureau
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[Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage, (continued)
- [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage, Kővágó, Zoltán, 2019/07/09
  - Re: [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage, Marc-André Lureau, 2019/07/10
- [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Kővágó, Zoltán, 2019/07/09
  - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Markus Armbruster, 2019/07/10
    - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Zoltán Kővágó, 2019/07/10
    - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Dr. David Alan Gilbert, 2019/07/11
    - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Markus Armbruster, 2019/07/11
    - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Zoltán Kővágó, 2019/07/11
  - Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio, Marc-André Lureau, 2019/07/10
- [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX, Kővágó, Zoltán, 2019/07/09
  - Re: [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX, Marc-André Lureau <=
- [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present, Kővágó, Zoltán, 2019/07/09
  - Re: [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present, Marc-André Lureau, 2019/07/10
    - Re: [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present, Zoltán Kővágó, 2019/07/11
- [Qemu-devel] [PATCH 11/11] audio: use size_t where makes sense, Kővágó, Zoltán, 2019/07/09
- Re: [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends, no-reply, 2019/07/09
- Re: [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends, no-reply, 2019/07/09
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