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

["Philippe Mathieu-Daudé" <f4bug@amsat.org>]

Hi Zoltán,

On 12/23/18 9:52 PM, Kővágó, Zoltán wrote:
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
>  audio/alsaaudio.c       |   8 +-
>  audio/audio.c           | 162 ++++++++++++++++++++--------------------
>  audio/audio.h           |   4 +-
>  audio/audio_int.h       |  22 +++---
>  audio/audio_template.h  |   6 +-
>  audio/mixeng.h          |  11 ++-
>  audio/ossaudio.c        |  18 ++---
>  audio/paaudio.c         |   8 +-
>  audio/rate_template.h   |   2 +-
>  audio/sdlaudio.c        |   3 +-
>  audio/wavaudio.c        |   4 +-
>  include/sysemu/replay.h |   4 +-
>  replay/replay-audio.c   |  16 ++--
>  13 files changed, 133 insertions(+), 135 deletions(-)
> 
> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
> index 19de7d01cb..69e7a3868c 100644
> --- a/audio/alsaaudio.c
> +++ b/audio/alsaaudio.c
> @@ -747,8 +747,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
>  
>      alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
>      if (!alsa->pcm_buf) {
> -        dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
> -               hw->samples, 1 << hw->info.shift);
> +        dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
> +              hw->samples, 1 << hw->info.shift);
>          alsa_anal_close1 (&handle);
>          return -1;
>      }
> @@ -849,8 +849,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>  
>      alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>      if (!alsa->pcm_buf) {
> -        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
> -               hw->samples, 1 << hw->info.shift);
> +        dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
> +              hw->samples, 1 << hw->info.shift);
>          alsa_anal_close1 (&handle);
>          return -1;
>      }
> diff --git a/audio/audio.c b/audio/audio.c
> index 1ea80ba6a7..27a8a31a64 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -530,10 +530,10 @@ static int audio_attach_capture (HWVoiceOut *hw)
>  /*
>   * Hard voice (capture)
>   */
> -static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
> +static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
>  {
>      SWVoiceIn *sw;
> -    int m = hw->total_samples_captured;
> +    size_t m = hw->total_samples_captured;
>  
>      for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
>          if (sw->active) {
> @@ -543,28 +543,28 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
>      return m;
>  }
>  
> -int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
> +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
>  {
> -    int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
> -    if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> -        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> +    size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
> +    if (audio_bug(__func__, live > hw->samples)) {
> +        dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
>          return 0;
>      }
>      return live;
>  }
>  
> -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> -                           int live, int pending)
> +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
> +                             size_t live, size_t pending)
>  {
> -    int left = hw->samples - pending;
> -    int len = MIN (left, live);
> -    int clipped = 0;
> +    size_t left = hw->samples - pending;
> +    size_t len = MIN (left, live);
> +    size_t clipped = 0;
>  
>      while (len) {
>          struct st_sample *src = hw->mix_buf + hw->rpos;
>          uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
> -        int samples_till_end_of_buf = hw->samples - hw->rpos;
> -        int samples_to_clip = MIN (len, samples_till_end_of_buf);
> +        size_t samples_till_end_of_buf = hw->samples - hw->rpos;
> +        size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
>  
>          hw->clip (dst, src, samples_to_clip);
>  
> @@ -578,14 +578,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
>  /*
>   * Soft voice (capture)
>   */
> -static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
> +static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
>  {
>      HWVoiceIn *hw = sw->hw;
> -    int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> -    int rpos;
> +    ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> +    ssize_t rpos;
>  
>      if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> -        dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> +        dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
>          return 0;
>      }
>  
> @@ -598,17 +598,17 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
>      }
>  }
>  
> -static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
> +static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
>  {
>      HWVoiceIn *hw = sw->hw;
> -    int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
> +    size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
>      struct st_sample *src, *dst = sw->buf;
>  
>      rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
>  
>      live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> -    if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> -        dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
> +    if (audio_bug(__func__, live > hw->samples)) {
> +        dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
>          return 0;
>      }
>  
> @@ -622,9 +622,9 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
>  
>      while (swlim) {
>          src = hw->conv_buf + rpos;
> -        isamp = hw->wpos - rpos;
> -        /* XXX: <= ? */
> -        if (isamp <= 0) {
> +        if (hw->wpos > rpos) {
> +            isamp = hw->wpos - rpos;
> +        } else {
>              isamp = hw->samples - rpos;
>          }
>  
> @@ -633,11 +633,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
>          }
>          osamp = swlim;
>  
> -        if (audio_bug(__func__, osamp < 0)) {
> -            dolog ("osamp=%d\n", osamp);
> -            return 0;
> -        }
> -
>          st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
>          swlim -= osamp;
>          rpos = (rpos + isamp) % hw->samples;
> @@ -658,10 +653,10 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
>  /*
>   * Hard voice (playback)
>   */
> -static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
> +static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
>  {
>      SWVoiceOut *sw;
> -    int m = INT_MAX;
> +    size_t m = SIZE_MAX;
>      int nb_live = 0;
>  
>      for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
> @@ -675,9 +670,9 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
>      return m;
>  }
>  
> -static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> +static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
>  {
> -    int smin;
> +    size_t smin;
>      int nb_live1;
>  
>      smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
> @@ -686,10 +681,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
>      }
>  
>      if (nb_live1) {
> -        int live = smin;
> +        size_t live = smin;
>  
> -        if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> -            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> +        if (audio_bug(__func__, live > hw->samples)) {
> +            dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
>              return 0;
>          }
>          return live;
> @@ -700,10 +695,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
>  /*
>   * Soft voice (playback)
>   */
> -static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
> +static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
>  {
> -    int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
> -    int ret = 0, pos = 0, total = 0;
> +    size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
> +    size_t ret = 0, pos = 0, total = 0;
>  
>      if (!sw) {
>          return size;
> @@ -712,8 +707,8 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
>      hwsamples = sw->hw->samples;
>  
>      live = sw->total_hw_samples_mixed;
> -    if (audio_bug(__func__, live < 0 || live > hwsamples)) {
> -        dolog ("live=%d hw->samples=%d\n", live, hwsamples);
> +    if (audio_bug(__func__, live > hwsamples)) {
> +        dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
>          return 0;
>      }
>  
> @@ -767,7 +762,7 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
>  
>  #ifdef DEBUG_OUT
>      dolog (
> -        "%s: write size %d ret %d total sw %d\n",
> +        "%s: write size %zu ret %zu total sw %zu\n",
>          SW_NAME (sw),
>          size >> sw->info.shift,
>          ret,
> @@ -846,7 +841,7 @@ static void audio_timer (void *opaque)
>  /*
>   * Public API
>   */
> -int AUD_write (SWVoiceOut *sw, void *buf, int size)
> +size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
>  {
>      if (!sw) {
>          /* XXX: Consider options */
> @@ -861,7 +856,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
>      return audio_pcm_sw_write(sw, buf, size);
>  }
>  
> -int AUD_read (SWVoiceIn *sw, void *buf, int size)
> +size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
>  {
>      if (!sw) {
>          /* XXX: Consider options */
> @@ -970,17 +965,17 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
>      }
>  }
>  
> -static int audio_get_avail (SWVoiceIn *sw)
> +static size_t audio_get_avail (SWVoiceIn *sw)
>  {
> -    int live;
> +    size_t live;
>  
>      if (!sw) {
>          return 0;
>      }
>  
>      live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
> -    if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
> -        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
> +    if (audio_bug(__func__, live > sw->hw->samples)) {
> +        dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
>          return 0;
>      }
>  
> @@ -993,9 +988,9 @@ static int audio_get_avail (SWVoiceIn *sw)
>      return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
>  }
>  
> -static int audio_get_free (SWVoiceOut *sw)
> +static size_t audio_get_free(SWVoiceOut *sw)
>  {
> -    int live, dead;
> +    size_t live, dead;
>  
>      if (!sw) {
>          return 0;
> @@ -1003,8 +998,8 @@ static int audio_get_free (SWVoiceOut *sw)
>  
>      live = sw->total_hw_samples_mixed;
>  
> -    if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
> -        dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
> +    if (audio_bug(__func__, live > sw->hw->samples)) {
> +        dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
>          return 0;
>      }
>  
> @@ -1019,9 +1014,10 @@ static int audio_get_free (SWVoiceOut *sw)
>      return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
>  }
>  
> -static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
> +static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
> +                                        size_t samples)
>  {
> -    int n;
> +    size_t n;
>  
>      if (hw->enabled) {
>          SWVoiceCap *sc;
> @@ -1032,17 +1028,17 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
>  
>              n = samples;
>              while (n) {
> -                int till_end_of_hw = hw->samples - rpos2;
> -                int to_write = MIN (till_end_of_hw, n);
> -                int bytes = to_write << hw->info.shift;
> -                int written;
> +                size_t till_end_of_hw = hw->samples - rpos2;
> +                size_t to_write = MIN(till_end_of_hw, n);
> +                size_t bytes = to_write << hw->info.shift;
> +                size_t written;
>  
>                  sw->buf = hw->mix_buf + rpos2;
>                  written = audio_pcm_sw_write (sw, NULL, bytes);
>                  if (written - bytes) {
> -                    dolog ("Could not mix %d bytes into a capture "
> -                           "buffer, mixed %d\n",
> -                           bytes, written);
> +                    dolog("Could not mix %zu bytes into a capture "
> +                          "buffer, mixed %zu\n",
> +                          bytes, written);
>                      break;
>                  }
>                  n -= to_write;
> @@ -1051,9 +1047,9 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
>          }
>      }
>  
> -    n = MIN (samples, hw->samples - rpos);
> -    mixeng_clear (hw->mix_buf + rpos, n);
> -    mixeng_clear (hw->mix_buf, samples - n);
> +    n = MIN(samples, hw->samples - rpos);
> +    mixeng_clear(hw->mix_buf + rpos, n);
> +    mixeng_clear(hw->mix_buf, samples - n);
>  }
>  
>  static void audio_run_out (AudioState *s)
> @@ -1062,16 +1058,16 @@ static void audio_run_out (AudioState *s)
>      SWVoiceOut *sw;
>  
>      while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
> -        int played;
> -        int live, free, nb_live, cleanup_required, prev_rpos;
> +        size_t played, live, prev_rpos, free;
> +        int nb_live, cleanup_required;
>  
>          live = audio_pcm_hw_get_live_out (hw, &nb_live);
>          if (!nb_live) {
>              live = 0;
>          }
>  
> -        if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> -            dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> +        if (audio_bug(__func__, live > hw->samples)) {
> +            dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
>              continue;
>          }
>  
> @@ -1106,13 +1102,13 @@ static void audio_run_out (AudioState *s)
>          played = hw->pcm_ops->run_out (hw, live);
>          replay_audio_out(&played);
>          if (audio_bug(__func__, hw->rpos >= hw->samples)) {
> -            dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
> -                   hw->rpos, hw->samples, played);
> +            dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
> +                  hw->rpos, hw->samples, played);
>              hw->rpos = 0;
>          }
>  
>  #ifdef DEBUG_OUT
> -        dolog ("played=%d\n", played);
> +        dolog("played=%zu\n", played);
>  #endif
>  
>          if (played) {
> @@ -1127,8 +1123,8 @@ static void audio_run_out (AudioState *s)
>              }
>  
>              if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
> -                dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
> -                       played, sw->total_hw_samples_mixed);
> +                dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
> +                      played, sw->total_hw_samples_mixed);
>                  played = sw->total_hw_samples_mixed;
>              }
>  
> @@ -1168,7 +1164,7 @@ static void audio_run_in (AudioState *s)
>  
>      while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
>          SWVoiceIn *sw;
> -        int captured = 0, min;
> +        size_t captured = 0, min;
>  
>          if (replay_mode != REPLAY_MODE_PLAY) {
>              captured = hw->pcm_ops->run_in(hw);
> @@ -1183,7 +1179,7 @@ static void audio_run_in (AudioState *s)
>              sw->total_hw_samples_acquired -= min;
>  
>              if (sw->active) {
> -                int avail;
> +                size_t avail;
>  
>                  avail = audio_get_avail (sw);
>                  if (avail > 0) {
> @@ -1199,15 +1195,15 @@ static void audio_run_capture (AudioState *s)
>      CaptureVoiceOut *cap;
>  
>      for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
> -        int live, rpos, captured;
> +        size_t live, rpos, captured;
>          HWVoiceOut *hw = &cap->hw;
>          SWVoiceOut *sw;
>  
>          captured = live = audio_pcm_hw_get_live_out (hw, NULL);
>          rpos = hw->rpos;
>          while (live) {
> -            int left = hw->samples - rpos;
> -            int to_capture = MIN (live, left);
> +            size_t left = hw->samples - rpos;
> +            size_t to_capture = MIN(live, left);
>              struct st_sample *src;
>              struct capture_callback *cb;
>  
> @@ -1230,8 +1226,8 @@ static void audio_run_capture (AudioState *s)
>              }
>  
>              if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
> -                dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
> -                       captured, sw->total_hw_samples_mixed);
> +                dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
> +                      captured, sw->total_hw_samples_mixed);
>                  captured = sw->total_hw_samples_mixed;
>              }
>  
> @@ -1570,8 +1566,8 @@ CaptureVoiceOut *AUD_add_capture(
>          hw->mix_buf = audio_calloc(__func__, hw->samples,
>                                     sizeof(struct st_sample));
>          if (!hw->mix_buf) {
> -            dolog ("Could not allocate capture mix buffer (%d samples)\n",
> -                   hw->samples);
> +            dolog("Could not allocate capture mix buffer (%zu samples)\n",
> +                  hw->samples);
>              goto err2;
>          }
>  
> @@ -1580,7 +1576,7 @@ CaptureVoiceOut *AUD_add_capture(
>          cap->buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>          if (!cap->buf) {
>              dolog ("Could not allocate capture buffer "
> -                   "(%d samples, each %d bytes)\n",
> +                   "(%zu samples, each %d bytes)\n",
>                     hw->samples, 1 << hw->info.shift);
>              goto err3;
>          }
> diff --git a/audio/audio.h b/audio/audio.h
> index bcbe56d639..bfb12e7531 100644
> --- a/audio/audio.h
> +++ b/audio/audio.h
> @@ -114,7 +114,7 @@ SWVoiceOut *AUD_open_out (
>      );
>  
>  void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
> -int  AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);

Can you add short documentation while modifying headers? Such:

/**
 * AUD_write:
 *
 * Returns: the number of bytes written.
 */

> +size_t AUD_write (SWVoiceOut *sw, void *pcm_buf, size_t size);
>  int  AUD_get_buffer_size_out (SWVoiceOut *sw);
>  void AUD_set_active_out (SWVoiceOut *sw, int on);
>  int  AUD_is_active_out (SWVoiceOut *sw);
> @@ -135,7 +135,7 @@ SWVoiceIn *AUD_open_in (
>      );
>  
>  void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
> -int  AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
> +size_t AUD_read (SWVoiceIn *sw, void *pcm_buf, size_t size);
>  void AUD_set_active_in (SWVoiceIn *sw, int on);
>  int  AUD_is_active_in (SWVoiceIn *sw);
>  
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index d269c38465..330c465d0b 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -60,12 +60,12 @@ typedef struct HWVoiceOut {
>  
>      f_sample *clip;
>  
> -    int rpos;
> +    size_t rpos;
>      uint64_t ts_helper;
>  
>      struct st_sample *mix_buf;
>  
> -    int samples;
> +    size_t samples;
>      QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
>      QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
>      int ctl_caps;
> @@ -81,13 +81,13 @@ typedef struct HWVoiceIn {
>  
>      t_sample *conv;
>  
> -    int wpos;
> -    int total_samples_captured;
> +    size_t wpos;
> +    size_t total_samples_captured;
>      uint64_t ts_helper;
>  
>      struct st_sample *conv_buf;
>  
> -    int samples;
> +    size_t samples;
>      QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
>      int ctl_caps;
>      struct audio_pcm_ops *pcm_ops;
> @@ -102,7 +102,7 @@ struct SWVoiceOut {
>      int64_t ratio;
>      struct st_sample *buf;
>      void *rate;
> -    int total_hw_samples_mixed;
> +    size_t total_hw_samples_mixed;
>      int active;
>      int empty;
>      HWVoiceOut *hw;
> @@ -119,7 +119,7 @@ struct SWVoiceIn {
>      struct audio_pcm_info info;
>      int64_t ratio;
>      void *rate;
> -    int total_hw_samples_acquired;
> +    size_t total_hw_samples_acquired;
>      struct st_sample *buf;
>      f_sample *clip;
>      HWVoiceIn *hw;

You forgot to update the prototypes in audio_pcm_ops.

> @@ -207,10 +207,10 @@ audio_driver *audio_driver_lookup(const char *name);
>  void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
>  void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
>  
> -int  audio_pcm_hw_get_live_in (HWVoiceIn *hw);
> +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
>  
> -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> -                           int live, int pending);
> +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
> +                             size_t live, size_t pending);
>  
>  int audio_bug (const char *funcname, int cond);
>  void *audio_calloc (const char *funcname, int nmemb, size_t size);
> @@ -223,7 +223,7 @@ void audio_run(AudioState *s, const char *msg);
>  
>  #define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
>  
> -static inline int audio_ring_dist (int dst, int src, int len)
> +static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
>  {
>      return (dst >= src) ? (dst - src) : (len - src + dst);
>  }
> diff --git a/audio/audio_template.h b/audio/audio_template.h
> index ce1e5d6559..fecbf1a046 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -79,8 +79,8 @@ static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)

Can you update this function to return a boolean?

>  {
>      HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
>      if (!HWBUF) {
> -        dolog ("Could not allocate " NAME " buffer (%d samples)\n",
> -               hw->samples);
> +        dolog("Could not allocate " NAME " buffer (%zu samples)\n",
> +              hw->samples);
>          return -1;
>      }
>  
> @@ -265,7 +265,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
>      }
>  
>      if (audio_bug(__func__, hw->samples <= 0)) {
> -        dolog ("hw->samples=%d\n", hw->samples);
> +        dolog("hw->samples=%zd\n", hw->samples);
>          goto err1;
>      }
>  
> diff --git a/audio/mixeng.h b/audio/mixeng.h
> index b53a5ef99a..2c09ed41e7 100644
> --- a/audio/mixeng.h
> +++ b/audio/mixeng.h
> @@ -25,6 +25,8 @@
>  #ifndef QEMU_MIXENG_H
>  #define QEMU_MIXENG_H
>  
> +#include <stddef.h>

This shouldn't be necessary, since source files include "qemu/osdep.h".

$ git grep -L qemu/osdep.h audio/*.c
$

> +
>  #ifdef FLOAT_MIXENG
>  typedef float mixeng_real;
>  struct mixeng_volume { int mute; mixeng_real r; mixeng_real l; };
> @@ -33,6 +35,7 @@ struct st_sample { mixeng_real l; mixeng_real r; };
>  struct mixeng_volume { int mute; int64_t r; int64_t l; };
>  struct st_sample { int64_t l; int64_t r; };
>  #endif
> +typedef struct st_sample st_sample;
>  
>  typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
>  typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
> @@ -41,10 +44,10 @@ extern t_sample *mixeng_conv[2][2][2][3];
>  extern f_sample *mixeng_clip[2][2][2][3];
>  
>  void *st_rate_start (int inrate, int outrate);
> -void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> -                   int *isamp, int *osamp);
> -void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> -                       int *isamp, int *osamp);
> +void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
> +                  size_t *isamp, size_t *osamp);
> +void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
> +                      size_t *isamp, size_t *osamp);
>  void st_rate_stop (void *opaque);
>  void mixeng_clear (struct st_sample *buf, int len);
>  void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
> diff --git a/audio/ossaudio.c b/audio/ossaudio.c
> index 70909e5696..05c7d6f85c 100644
> --- a/audio/ossaudio.c
> +++ b/audio/ossaudio.c
> @@ -476,8 +476,8 @@ static void oss_fini_out (HWVoiceOut *hw)
>          if (oss->mmapped) {
>              err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
>              if (err) {
> -                oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
> -                            oss->pcm_buf, hw->samples << hw->info.shift);
> +                oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
> +                           oss->pcm_buf, hw->samples << hw->info.shift);
>              }
>          }
>          else {
> @@ -543,8 +543,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
>              0
>              );
>          if (oss->pcm_buf == MAP_FAILED) {
> -            oss_logerr (errno, "Failed to map %d bytes of DAC\n",
> -                        hw->samples << hw->info.shift);
> +            oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
> +                       hw->samples << hw->info.shift);
>          }
>          else {
>              int err;
> @@ -568,8 +568,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
>              if (!oss->mmapped) {
>                  err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
>                  if (err) {
> -                    oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
> -                                oss->pcm_buf, hw->samples << hw->info.shift);
> +                    oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
> +                               oss->pcm_buf, hw->samples << hw->info.shift);
>                  }
>              }
>          }
> @@ -581,7 +581,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
>                                      1 << hw->info.shift);
>          if (!oss->pcm_buf) {
>              dolog (
> -                "Could not allocate DAC buffer (%d samples, each %d bytes)\n",
> +                "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
>                  hw->samples,
>                  1 << hw->info.shift
>                  );
> @@ -693,8 +693,8 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>      hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
>      oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>      if (!oss->pcm_buf) {
> -        dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
> -               hw->samples, 1 << hw->info.shift);
> +        dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
> +              hw->samples, 1 << hw->info.shift);
>          oss_anal_close (&fd);
>          return -1;
>      }
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index 6a1919e93b..251b087a74 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -584,8 +584,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
>      pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>      pa->rpos = hw->rpos;
>      if (!pa->pcm_buf) {
> -        dolog ("Could not allocate buffer (%d bytes)\n",
> -               hw->samples << hw->info.shift);
> +        dolog("Could not allocate buffer (%zu bytes)\n",
> +              hw->samples << hw->info.shift);
>          goto fail2;
>      }
>  
> @@ -645,8 +645,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>      pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>      pa->wpos = hw->wpos;
>      if (!pa->pcm_buf) {
> -        dolog ("Could not allocate buffer (%d bytes)\n",
> -               hw->samples << hw->info.shift);
> +        dolog("Could not allocate buffer (%zu bytes)\n",
> +              hw->samples << hw->info.shift);
>          goto fail2;
>      }
>  
> diff --git a/audio/rate_template.h b/audio/rate_template.h
> index 6e93588877..f94c940c61 100644
> --- a/audio/rate_template.h
> +++ b/audio/rate_template.h
> @@ -28,7 +28,7 @@
>   * Return number of samples processed.
>   */
>  void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> -           int *isamp, int *osamp)
> +           size_t *isamp, size_t *osamp)
>  {
>      struct rate *rate = opaque;
>      struct st_sample *istart, *iend;
> diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
> index ab9166d054..92da4804c6 100644
> --- a/audio/sdlaudio.c
> +++ b/audio/sdlaudio.c
> @@ -273,8 +273,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
>          }
>  
>          if (audio_bug(__func__, sdl->live < 0 || sdl->live > hw->samples)) {
> -            dolog ("sdl->live=%d hw->samples=%d\n",
> -                   sdl->live, hw->samples);
> +            dolog("sdl->live=%d hw->samples=%zu\n", sdl->live, hw->samples);
>              return;
>          }
>  
> diff --git a/audio/wavaudio.c b/audio/wavaudio.c
> index dda6993fb9..58300663ff 100644
> --- a/audio/wavaudio.c
> +++ b/audio/wavaudio.c
> @@ -137,8 +137,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
>      hw->samples = 1024;
>      wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
>      if (!wav->pcm_buf) {
> -        dolog ("Could not allocate buffer (%d bytes)\n",
> -               hw->samples << hw->info.shift);
> +        dolog("Could not allocate buffer (%zu bytes)\n",
> +              hw->samples << hw->info.shift);
>          return -1;
>      }
>  
> diff --git a/include/sysemu/replay.h b/include/sysemu/replay.h
> index 3a7c58e423..5c0a91e44f 100644
> --- a/include/sysemu/replay.h
> +++ b/include/sysemu/replay.h
> @@ -179,9 +179,9 @@ void replay_net_packet_event(ReplayNetState *rns, unsigned flags,
>  /* Audio */
>  
>  /*! Saves/restores number of played samples of audio out operation. */
> -void replay_audio_out(int *played);
> +void replay_audio_out(size_t *played);
>  /*! Saves/restores recorded samples of audio in operation. */
> -void replay_audio_in(int *recorded, void *samples, int *wpos, int size);
> +void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size);
>  
>  /* VM state operations */
>  
> diff --git a/replay/replay-audio.c b/replay/replay-audio.c
> index b113836de4..efe1628727 100644
> --- a/replay/replay-audio.c
> +++ b/replay/replay-audio.c
> @@ -16,18 +16,18 @@
>  #include "sysemu/sysemu.h"
>  #include "audio/audio.h"
>  
> -void replay_audio_out(int *played)
> +void replay_audio_out(size_t *played)
>  {
>      if (replay_mode == REPLAY_MODE_RECORD) {
>          g_assert(replay_mutex_locked());
>          replay_save_instructions();
>          replay_put_event(EVENT_AUDIO_OUT);
> -        replay_put_dword(*played);
> +        replay_put_qword(*played);
>      } else if (replay_mode == REPLAY_MODE_PLAY) {
>          g_assert(replay_mutex_locked());
>          replay_account_executed_instructions();
>          if (replay_next_event_is(EVENT_AUDIO_OUT)) {
> -            *played = replay_get_dword();
> +            *played = replay_get_qword();
>              replay_finish_event();
>          } else {
>              error_report("Missing audio out event in the replay log");
> @@ -36,7 +36,7 @@ void replay_audio_out(int *played)
>      }
>  }
>  
> -void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> +void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size)
>  {
>      int pos;
>      uint64_t left, right;
> @@ -44,8 +44,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
>          g_assert(replay_mutex_locked());
>          replay_save_instructions();
>          replay_put_event(EVENT_AUDIO_IN);
> -        replay_put_dword(*recorded);
> -        replay_put_dword(*wpos);
> +        replay_put_qword(*recorded);
> +        replay_put_qword(*wpos);
>          for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
>               ; pos = (pos + 1) % size) {
>              audio_sample_to_uint64(samples, pos, &left, &right);
> @@ -56,8 +56,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
>          g_assert(replay_mutex_locked());
>          replay_account_executed_instructions();
>          if (replay_next_event_is(EVENT_AUDIO_IN)) {
> -            *recorded = replay_get_dword();
> -            *wpos = replay_get_dword();
> +            *recorded = replay_get_qword();
> +            *wpos = replay_get_qword();
>              for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
>                   ; pos = (pos + 1) % size) {
>                  left = replay_get_qword();
> 

Except the audio_pcm_ops prototypes not using size_t, and Pavel comment
about replay version update, this patch looks sane to me.
Returning size_t seems enough (no need to return a ssize_t).

Regards,

Phil.