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Re: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
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From: |
Philippe Mathieu-Daudé |
|
Subject: |
Re: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense |
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Date: |
Tue, 25 Dec 2018 12:08:40 +0100 |
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User-agent: |
Mozilla/5.0 (X11; Linux x86_64; rv:60.0) Gecko/20100101 Thunderbird/60.3.1 |
Hi Zoltán,
On 12/23/18 9:52 PM, Kővágó, Zoltán wrote:
> Signed-off-by: Kővágó, Zoltán <address@hidden>
> ---
> 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.
- [Qemu-devel] [PATCH v2 22/52] paaudio: properly disconnect streams in fini_*, (continued)
- [Qemu-devel] [PATCH v2 22/52] paaudio: properly disconnect streams in fini_*, Kővágó, Zoltán, 2018/12/23
- [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, 2018/12/23
- 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