* [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:59 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio Kővágó, Zoltán
` (11 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Remove glob_audio_state from functions, where possible without breaking
the API. This means that most static functions in audio.c now take an
AudioState pointer instead of implicitly using glob_audio_state. Also
included a pointer in SWVoice*, HWVoice* structs, so that functions
dealing them can know the audio state without having to pass it around
separately.
This is required in order to support multiple simultaneous audio
backends (added in a later commit).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_int.h | 8 ++++++
audio/audio_template.h | 46 ++++++++++++++++----------------
audio/audio.c | 59 +++++++++++++++++++-----------------------
3 files changed, 57 insertions(+), 56 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 3f14842709..8164696b2c 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -49,9 +49,11 @@ struct audio_pcm_info {
int swap_endianness;
};
+typedef struct AudioState AudioState;
typedef struct SWVoiceCap SWVoiceCap;
typedef struct HWVoiceOut {
+ AudioState *s;
int enabled;
int poll_mode;
int pending_disable;
@@ -73,6 +75,7 @@ typedef struct HWVoiceOut {
} HWVoiceOut;
typedef struct HWVoiceIn {
+ AudioState *s;
int enabled;
int poll_mode;
struct audio_pcm_info info;
@@ -94,6 +97,7 @@ typedef struct HWVoiceIn {
struct SWVoiceOut {
QEMUSoundCard *card;
+ AudioState *s;
struct audio_pcm_info info;
t_sample *conv;
int64_t ratio;
@@ -111,6 +115,7 @@ struct SWVoiceOut {
struct SWVoiceIn {
QEMUSoundCard *card;
+ AudioState *s;
int active;
struct audio_pcm_info info;
int64_t ratio;
@@ -188,6 +193,9 @@ typedef struct AudioState {
int nb_hw_voices_in;
int vm_running;
int64_t period_ticks;
+
+ bool timer_running;
+ uint64_t timer_last;
} AudioState;
extern const struct mixeng_volume nominal_volume;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 1232bb54db..c721fed75d 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -36,9 +36,9 @@
#define HWBUF hw->conv_buf
#endif
-static void glue (audio_init_nb_voices_, TYPE) (struct audio_driver *drv)
+static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
+ struct audio_driver *drv)
{
- AudioState *s = &glob_audio_state;
int max_voices = glue (drv->max_voices_, TYPE);
int voice_size = glue (drv->voice_size_, TYPE);
@@ -183,8 +183,8 @@ static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
{
- AudioState *s = &glob_audio_state;
HW *hw = *hwp;
+ AudioState *s = hw->s;
if (!hw->sw_head.lh_first) {
#ifdef DAC
@@ -199,15 +199,14 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
}
}
-static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+static HW *glue(audio_pcm_hw_find_any_, TYPE)(AudioState *s, HW *hw)
{
- AudioState *s = &glob_audio_state;
return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first;
}
-static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
+static HW *glue(audio_pcm_hw_find_any_enabled_, TYPE)(AudioState *s, HW *hw)
{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (hw->enabled) {
return hw;
}
@@ -215,12 +214,10 @@ static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
return NULL;
}
-static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
- HW *hw,
- struct audsettings *as
- )
+static HW *glue(audio_pcm_hw_find_specific_, TYPE)(AudioState *s, HW *hw,
+ struct audsettings *as)
{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (audio_pcm_info_eq (&hw->info, as)) {
return hw;
}
@@ -228,10 +225,10 @@ static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
return NULL;
}
-static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
+static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
+ struct audsettings *as)
{
HW *hw;
- AudioState *s = &glob_audio_state;
struct audio_driver *drv = s->drv;
if (!glue (s->nb_hw_voices_, TYPE)) {
@@ -255,6 +252,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
return NULL;
}
+ hw->s = s;
hw->pcm_ops = drv->pcm_ops;
hw->ctl_caps = drv->ctl_caps;
@@ -328,33 +326,33 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
abort();
}
-static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
+static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
{
HW *hw;
- AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (pdo->fixed_settings) {
- hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+ hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (hw) {
return hw;
}
}
- hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, as);
+ hw = glue(audio_pcm_hw_find_specific_, TYPE)(s, NULL, as);
if (hw) {
return hw;
}
- hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+ hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (hw) {
return hw;
}
- return glue (audio_pcm_hw_find_any_, TYPE) (NULL);
+ return glue(audio_pcm_hw_find_any_, TYPE)(s, NULL);
}
-static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
+ AudioState *s,
const char *sw_name,
struct audsettings *as
)
@@ -362,7 +360,6 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
- AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (pdo->fixed_settings) {
@@ -378,8 +375,9 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
sw_name ? sw_name : "unknown", sizeof (*sw));
goto err1;
}
+ sw->s = s;
- hw = glue (audio_pcm_hw_add_, TYPE) (&hw_as);
+ hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
if (!hw) {
goto err2;
}
@@ -476,7 +474,7 @@ SW *glue (AUD_open_, TYPE) (
}
}
else {
- sw = glue (audio_pcm_create_voice_pair_, TYPE) (name, as);
+ sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
if (!sw) {
dolog ("Failed to create voice `%s'\n", name);
return NULL;
diff --git a/audio/audio.c b/audio/audio.c
index 05adf7ffeb..8d2f580788 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -399,12 +399,10 @@ static void noop_conv (struct st_sample *dst, const void *src, int samples)
(void) samples;
}
-static CaptureVoiceOut *audio_pcm_capture_find_specific (
- struct audsettings *as
- )
+static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
+ struct audsettings *as)
{
CaptureVoiceOut *cap;
- AudioState *s = &glob_audio_state;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
if (audio_pcm_info_eq (&cap->hw.info, as)) {
@@ -481,7 +479,7 @@ static void audio_detach_capture (HWVoiceOut *hw)
static int audio_attach_capture (HWVoiceOut *hw)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s = hw->s;
CaptureVoiceOut *cap;
audio_detach_capture (hw);
@@ -789,19 +787,15 @@ static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
/*
* Timer
*/
-
-static bool audio_timer_running;
-static uint64_t audio_timer_last;
-
-static int audio_is_timer_needed (void)
+static int audio_is_timer_needed(AudioState *s)
{
HWVoiceIn *hwi = NULL;
HWVoiceOut *hwo = NULL;
- while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
if (!hwo->poll_mode) return 1;
}
- while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
if (!hwi->poll_mode) return 1;
}
return 0;
@@ -809,18 +803,18 @@ static int audio_is_timer_needed (void)
static void audio_reset_timer (AudioState *s)
{
- if (audio_is_timer_needed ()) {
+ if (audio_is_timer_needed(s)) {
timer_mod_anticipate_ns(s->ts,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
- if (!audio_timer_running) {
- audio_timer_running = true;
- audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ if (!s->timer_running) {
+ s->timer_running = true;
+ s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
- if (audio_timer_running) {
- audio_timer_running = false;
+ if (s->timer_running) {
+ s->timer_running = false;
trace_audio_timer_stop();
}
}
@@ -832,11 +826,11 @@ static void audio_timer (void *opaque)
AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- diff = now - audio_timer_last;
+ diff = now - s->timer_last;
if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
- audio_timer_last = now;
+ s->timer_last = now;
audio_run("timer");
audio_reset_timer(s);
@@ -890,7 +884,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
hw = sw->hw;
if (sw->active != on) {
- AudioState *s = &glob_audio_state;
+ AudioState *s = sw->s;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
@@ -937,7 +931,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
hw = sw->hw;
if (sw->active != on) {
- AudioState *s = &glob_audio_state;
+ AudioState *s = sw->s;
SWVoiceIn *temp_sw;
if (on) {
@@ -1060,7 +1054,7 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
- while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
int played;
int live, free, nb_live, cleanup_required, prev_rpos;
@@ -1165,7 +1159,7 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
- while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
int captured = 0, min;
@@ -1271,8 +1265,8 @@ static int audio_driver_init(AudioState *s, struct audio_driver *drv,
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
- audio_init_nb_voices_out (drv);
- audio_init_nb_voices_in (drv);
+ audio_init_nb_voices_out(s, drv);
+ audio_init_nb_voices_in(s, drv);
s->drv = drv;
return 0;
}
@@ -1293,11 +1287,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
int op = running ? VOICE_ENABLE : VOICE_DISABLE;
s->vm_running = running;
- while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
hwo->pcm_ops->ctl_out(hwo, op);
}
- while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
hwi->pcm_ops->ctl_in(hwi, op);
}
audio_reset_timer (s);
@@ -1317,7 +1311,7 @@ void audio_cleanup(void)
HWVoiceIn *hwi, *hwin;
is_cleaning_up = true;
- QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) {
+ QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
if (hwo->enabled) {
@@ -1336,7 +1330,7 @@ void audio_cleanup(void)
QLIST_REMOVE(hwo, entries);
}
- QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) {
+ QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
if (hwi->enabled) {
hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
}
@@ -1532,7 +1526,7 @@ CaptureVoiceOut *AUD_add_capture (
cb->ops = *ops;
cb->opaque = cb_opaque;
- cap = audio_pcm_capture_find_specific (as);
+ cap = audio_pcm_capture_find_specific(s, as);
if (cap) {
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return cap;
@@ -1544,6 +1538,7 @@ CaptureVoiceOut *AUD_add_capture (
cap = g_malloc0(sizeof(*cap));
hw = &cap->hw;
+ hw->s = s;
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
@@ -1564,7 +1559,7 @@ CaptureVoiceOut *AUD_add_capture (
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
- QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) {
+ QLIST_FOREACH(hw, &s->hw_head_out, entries) {
audio_attach_capture (hw);
}
return cap;
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage
2019-07-09 18:35 ` [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage Kővágó, Zoltán
@ 2019-07-10 19:59 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:59 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:57 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Remove glob_audio_state from functions, where possible without breaking
> the API. This means that most static functions in audio.c now take an
> AudioState pointer instead of implicitly using glob_audio_state. Also
> included a pointer in SWVoice*, HWVoice* structs, so that functions
> dealing them can know the audio state without having to pass it around
> separately.
>
> This is required in order to support multiple simultaneous audio
> backends (added in a later commit).
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> audio/audio_int.h | 8 ++++++
> audio/audio_template.h | 46 ++++++++++++++++----------------
> audio/audio.c | 59 +++++++++++++++++++-----------------------
> 3 files changed, 57 insertions(+), 56 deletions(-)
>
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index 3f14842709..8164696b2c 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -49,9 +49,11 @@ struct audio_pcm_info {
> int swap_endianness;
> };
>
> +typedef struct AudioState AudioState;
> typedef struct SWVoiceCap SWVoiceCap;
>
> typedef struct HWVoiceOut {
> + AudioState *s;
> int enabled;
> int poll_mode;
> int pending_disable;
> @@ -73,6 +75,7 @@ typedef struct HWVoiceOut {
> } HWVoiceOut;
>
> typedef struct HWVoiceIn {
> + AudioState *s;
> int enabled;
> int poll_mode;
> struct audio_pcm_info info;
> @@ -94,6 +97,7 @@ typedef struct HWVoiceIn {
>
> struct SWVoiceOut {
> QEMUSoundCard *card;
> + AudioState *s;
> struct audio_pcm_info info;
> t_sample *conv;
> int64_t ratio;
> @@ -111,6 +115,7 @@ struct SWVoiceOut {
>
> struct SWVoiceIn {
> QEMUSoundCard *card;
> + AudioState *s;
> int active;
> struct audio_pcm_info info;
> int64_t ratio;
> @@ -188,6 +193,9 @@ typedef struct AudioState {
> int nb_hw_voices_in;
> int vm_running;
> int64_t period_ticks;
> +
> + bool timer_running;
> + uint64_t timer_last;
> } AudioState;
>
> extern const struct mixeng_volume nominal_volume;
> diff --git a/audio/audio_template.h b/audio/audio_template.h
> index 1232bb54db..c721fed75d 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -36,9 +36,9 @@
> #define HWBUF hw->conv_buf
> #endif
>
> -static void glue (audio_init_nb_voices_, TYPE) (struct audio_driver *drv)
> +static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
> + struct audio_driver *drv)
> {
> - AudioState *s = &glob_audio_state;
> int max_voices = glue (drv->max_voices_, TYPE);
> int voice_size = glue (drv->voice_size_, TYPE);
>
> @@ -183,8 +183,8 @@ static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
>
> static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
> {
> - AudioState *s = &glob_audio_state;
> HW *hw = *hwp;
> + AudioState *s = hw->s;
>
> if (!hw->sw_head.lh_first) {
> #ifdef DAC
> @@ -199,15 +199,14 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
> }
> }
>
> -static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
> +static HW *glue(audio_pcm_hw_find_any_, TYPE)(AudioState *s, HW *hw)
> {
> - AudioState *s = &glob_audio_state;
> return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first;
> }
>
> -static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
> +static HW *glue(audio_pcm_hw_find_any_enabled_, TYPE)(AudioState *s, HW *hw)
> {
> - while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
> + while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
> if (hw->enabled) {
> return hw;
> }
> @@ -215,12 +214,10 @@ static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
> return NULL;
> }
>
> -static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
> - HW *hw,
> - struct audsettings *as
> - )
> +static HW *glue(audio_pcm_hw_find_specific_, TYPE)(AudioState *s, HW *hw,
> + struct audsettings *as)
> {
> - while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
> + while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
> if (audio_pcm_info_eq (&hw->info, as)) {
> return hw;
> }
> @@ -228,10 +225,10 @@ static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
> return NULL;
> }
>
> -static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
> +static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
> + struct audsettings *as)
> {
> HW *hw;
> - AudioState *s = &glob_audio_state;
> struct audio_driver *drv = s->drv;
>
> if (!glue (s->nb_hw_voices_, TYPE)) {
> @@ -255,6 +252,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
> return NULL;
> }
>
> + hw->s = s;
> hw->pcm_ops = drv->pcm_ops;
> hw->ctl_caps = drv->ctl_caps;
>
> @@ -328,33 +326,33 @@ AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
> abort();
> }
>
> -static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
> +static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
> {
> HW *hw;
> - AudioState *s = &glob_audio_state;
> AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
>
> if (pdo->fixed_settings) {
> - hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
> + hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
> if (hw) {
> return hw;
> }
> }
>
> - hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, as);
> + hw = glue(audio_pcm_hw_find_specific_, TYPE)(s, NULL, as);
> if (hw) {
> return hw;
> }
>
> - hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
> + hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
> if (hw) {
> return hw;
> }
>
> - return glue (audio_pcm_hw_find_any_, TYPE) (NULL);
> + return glue(audio_pcm_hw_find_any_, TYPE)(s, NULL);
> }
>
> -static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
> +static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
> + AudioState *s,
> const char *sw_name,
> struct audsettings *as
> )
> @@ -362,7 +360,6 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
> SW *sw;
> HW *hw;
> struct audsettings hw_as;
> - AudioState *s = &glob_audio_state;
> AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
>
> if (pdo->fixed_settings) {
> @@ -378,8 +375,9 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
> sw_name ? sw_name : "unknown", sizeof (*sw));
> goto err1;
> }
> + sw->s = s;
>
> - hw = glue (audio_pcm_hw_add_, TYPE) (&hw_as);
> + hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
> if (!hw) {
> goto err2;
> }
> @@ -476,7 +474,7 @@ SW *glue (AUD_open_, TYPE) (
> }
> }
> else {
> - sw = glue (audio_pcm_create_voice_pair_, TYPE) (name, as);
> + sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
> if (!sw) {
> dolog ("Failed to create voice `%s'\n", name);
> return NULL;
> diff --git a/audio/audio.c b/audio/audio.c
> index 05adf7ffeb..8d2f580788 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -399,12 +399,10 @@ static void noop_conv (struct st_sample *dst, const void *src, int samples)
> (void) samples;
> }
>
> -static CaptureVoiceOut *audio_pcm_capture_find_specific (
> - struct audsettings *as
> - )
> +static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
> + struct audsettings *as)
> {
> CaptureVoiceOut *cap;
> - AudioState *s = &glob_audio_state;
>
> for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
> if (audio_pcm_info_eq (&cap->hw.info, as)) {
> @@ -481,7 +479,7 @@ static void audio_detach_capture (HWVoiceOut *hw)
>
> static int audio_attach_capture (HWVoiceOut *hw)
> {
> - AudioState *s = &glob_audio_state;
> + AudioState *s = hw->s;
> CaptureVoiceOut *cap;
>
> audio_detach_capture (hw);
> @@ -789,19 +787,15 @@ static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
> /*
> * Timer
> */
> -
> -static bool audio_timer_running;
> -static uint64_t audio_timer_last;
> -
> -static int audio_is_timer_needed (void)
> +static int audio_is_timer_needed(AudioState *s)
> {
> HWVoiceIn *hwi = NULL;
> HWVoiceOut *hwo = NULL;
>
> - while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
> + while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
> if (!hwo->poll_mode) return 1;
> }
> - while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
> + while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
> if (!hwi->poll_mode) return 1;
> }
> return 0;
> @@ -809,18 +803,18 @@ static int audio_is_timer_needed (void)
>
> static void audio_reset_timer (AudioState *s)
> {
> - if (audio_is_timer_needed ()) {
> + if (audio_is_timer_needed(s)) {
> timer_mod_anticipate_ns(s->ts,
> qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
> - if (!audio_timer_running) {
> - audio_timer_running = true;
> - audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
> + if (!s->timer_running) {
> + s->timer_running = true;
> + s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
> trace_audio_timer_start(s->period_ticks / SCALE_MS);
> }
> } else {
> timer_del(s->ts);
> - if (audio_timer_running) {
> - audio_timer_running = false;
> + if (s->timer_running) {
> + s->timer_running = false;
> trace_audio_timer_stop();
> }
> }
> @@ -832,11 +826,11 @@ static void audio_timer (void *opaque)
> AudioState *s = opaque;
>
> now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
> - diff = now - audio_timer_last;
> + diff = now - s->timer_last;
> if (diff > s->period_ticks * 3 / 2) {
> trace_audio_timer_delayed(diff / SCALE_MS);
> }
> - audio_timer_last = now;
> + s->timer_last = now;
>
> audio_run("timer");
> audio_reset_timer(s);
> @@ -890,7 +884,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
>
> hw = sw->hw;
> if (sw->active != on) {
> - AudioState *s = &glob_audio_state;
> + AudioState *s = sw->s;
> SWVoiceOut *temp_sw;
> SWVoiceCap *sc;
>
> @@ -937,7 +931,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
>
> hw = sw->hw;
> if (sw->active != on) {
> - AudioState *s = &glob_audio_state;
> + AudioState *s = sw->s;
> SWVoiceIn *temp_sw;
>
> if (on) {
> @@ -1060,7 +1054,7 @@ static void audio_run_out (AudioState *s)
> HWVoiceOut *hw = NULL;
> SWVoiceOut *sw;
>
> - while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
> + while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
> int played;
> int live, free, nb_live, cleanup_required, prev_rpos;
>
> @@ -1165,7 +1159,7 @@ static void audio_run_in (AudioState *s)
> {
> HWVoiceIn *hw = NULL;
>
> - while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
> + while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
> SWVoiceIn *sw;
> int captured = 0, min;
>
> @@ -1271,8 +1265,8 @@ static int audio_driver_init(AudioState *s, struct audio_driver *drv,
> s->drv_opaque = drv->init(dev);
>
> if (s->drv_opaque) {
> - audio_init_nb_voices_out (drv);
> - audio_init_nb_voices_in (drv);
> + audio_init_nb_voices_out(s, drv);
> + audio_init_nb_voices_in(s, drv);
> s->drv = drv;
> return 0;
> }
> @@ -1293,11 +1287,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
> int op = running ? VOICE_ENABLE : VOICE_DISABLE;
>
> s->vm_running = running;
> - while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
> + while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
> hwo->pcm_ops->ctl_out(hwo, op);
> }
>
> - while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
> + while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
> hwi->pcm_ops->ctl_in(hwi, op);
> }
> audio_reset_timer (s);
> @@ -1317,7 +1311,7 @@ void audio_cleanup(void)
> HWVoiceIn *hwi, *hwin;
>
> is_cleaning_up = true;
> - QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) {
> + QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
> SWVoiceCap *sc;
>
> if (hwo->enabled) {
> @@ -1336,7 +1330,7 @@ void audio_cleanup(void)
> QLIST_REMOVE(hwo, entries);
> }
>
> - QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) {
> + QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
> if (hwi->enabled) {
> hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
> }
> @@ -1532,7 +1526,7 @@ CaptureVoiceOut *AUD_add_capture (
> cb->ops = *ops;
> cb->opaque = cb_opaque;
>
> - cap = audio_pcm_capture_find_specific (as);
> + cap = audio_pcm_capture_find_specific(s, as);
> if (cap) {
> QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
> return cap;
> @@ -1544,6 +1538,7 @@ CaptureVoiceOut *AUD_add_capture (
> cap = g_malloc0(sizeof(*cap));
>
> hw = &cap->hw;
> + hw->s = s;
> QLIST_INIT (&hw->sw_head);
> QLIST_INIT (&cap->cb_head);
>
> @@ -1564,7 +1559,7 @@ CaptureVoiceOut *AUD_add_capture (
> QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
> QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
>
> - QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) {
> + QLIST_FOREACH(hw, &s->hw_head_out, entries) {
> audio_attach_capture (hw);
> }
> return cap;
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
2019-07-09 18:35 ` [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 4:06 ` Markus Armbruster
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 03/11] audio: add audiodev properties to frontends Kővágó, Zoltán
` (10 subsequent siblings)
12 siblings, 2 replies; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Markus Armbruster, Gerd Hoffmann, Dr. David Alan Gilbert
Audio functions no longer access glob_audio_state, instead they get an
AudioState as a parameter. This is required in order to support
multiple backends.
glob_audio_state is also gone, and replaced with a tailq so we can store
more than one states.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.h | 12 +++--
audio/audio_int.h | 2 +
audio/audio_template.h | 2 +-
ui/vnc.h | 2 +
audio/audio.c | 102 +++++++++++++++++++++++++++++++----------
audio/wavcapture.c | 6 +--
monitor/misc.c | 12 ++++-
ui/vnc.c | 15 +++++-
hmp-commands.hx | 11 +++--
qemu-options.hx | 5 ++
10 files changed, 131 insertions(+), 38 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index 64b0f761bc..ad2457f4de 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -78,8 +78,10 @@ typedef struct SWVoiceOut SWVoiceOut;
typedef struct CaptureVoiceOut CaptureVoiceOut;
typedef struct SWVoiceIn SWVoiceIn;
+typedef struct AudioState AudioState;
typedef struct QEMUSoundCard {
char *name;
+ AudioState *state;
QLIST_ENTRY (QEMUSoundCard) entries;
} QEMUSoundCard;
@@ -92,7 +94,8 @@ void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
void AUD_register_card (const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
-CaptureVoiceOut *AUD_add_capture (
+CaptureVoiceOut *AUD_add_capture(
+ AudioState *s,
struct audsettings *as,
struct audio_capture_ops *ops,
void *opaque
@@ -160,8 +163,8 @@ static inline void *advance (void *p, int incr)
#define audio_MAX(a, b) ((a)<(b)?(b):(a))
#endif
-int wav_start_capture (CaptureState *s, const char *path, int freq,
- int bits, int nchannels);
+int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
+ int freq, int bits, int nchannels);
bool audio_is_cleaning_up(void);
void audio_cleanup(void);
@@ -175,4 +178,7 @@ void audio_parse_option(const char *opt);
void audio_init_audiodevs(void);
void audio_legacy_help(void);
+AudioState *audio_state_by_name(const char *name);
+const char *audio_get_id(QEMUSoundCard *card);
+
#endif /* QEMU_AUDIO_H */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 8164696b2c..9f01f6ad00 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -196,6 +196,8 @@ typedef struct AudioState {
bool timer_running;
uint64_t timer_last;
+
+ QTAILQ_ENTRY(AudioState) list;
} AudioState;
extern const struct mixeng_volume nominal_volume;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index c721fed75d..54f07338e7 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -428,7 +428,7 @@ SW *glue (AUD_open_, TYPE) (
struct audsettings *as
)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s = card->state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
diff --git a/ui/vnc.h b/ui/vnc.h
index 2f84db3142..6f54653455 100644
--- a/ui/vnc.h
+++ b/ui/vnc.h
@@ -183,6 +183,8 @@ struct VncDisplay
#ifdef CONFIG_VNC_SASL
VncDisplaySASL sasl;
#endif
+
+ AudioState *audio_state;
};
typedef struct VncTight {
diff --git a/audio/audio.c b/audio/audio.c
index 8d2f580788..e9dd7c8b32 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -87,7 +87,8 @@ audio_driver *audio_driver_lookup(const char *name)
return NULL;
}
-static AudioState glob_audio_state;
+static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
+ QTAILQ_HEAD_INITIALIZER(audio_states);
const struct mixeng_volume nominal_volume = {
.mute = 0,
@@ -1236,11 +1237,14 @@ static void audio_run_capture (AudioState *s)
void audio_run (const char *msg)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s;
+
+ QTAILQ_FOREACH(s, &audio_states, list) {
+ audio_run_out(s);
+ audio_run_in(s);
+ audio_run_capture(s);
+ }
- audio_run_out (s);
- audio_run_in (s);
- audio_run_capture (s);
#ifdef DEBUG_POLL
{
static double prevtime;
@@ -1304,13 +1308,11 @@ bool audio_is_cleaning_up(void)
return is_cleaning_up;
}
-void audio_cleanup(void)
+static void free_audio_state(AudioState *s)
{
- AudioState *s = &glob_audio_state;
HWVoiceOut *hwo, *hwon;
HWVoiceIn *hwi, *hwin;
- is_cleaning_up = true;
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
@@ -1347,6 +1349,17 @@ void audio_cleanup(void)
qapi_free_Audiodev(s->dev);
s->dev = NULL;
}
+ g_free(s);
+}
+
+void audio_cleanup(void)
+{
+ is_cleaning_up = true;
+ while (!QTAILQ_EMPTY(&audio_states)) {
+ AudioState *s = QTAILQ_FIRST(&audio_states);
+ QTAILQ_REMOVE(&audio_states, s, list);
+ free_audio_state(s);
+ }
}
static const VMStateDescription vmstate_audio = {
@@ -1373,28 +1386,33 @@ static AudiodevListEntry *audiodev_find(
return NULL;
}
-static int audio_init(Audiodev *dev)
+/*
+ * if we have dev, this function was called because of an -audiodev argument =>
+ * initialize a new state with it
+ * if dev == NULL => legacy implicit initialization, return the already created
+ * state or create a new one
+ */
+static AudioState *audio_init(Audiodev *dev)
{
+ static bool atexit_registered;
size_t i;
int done = 0;
const char *drvname = NULL;
VMChangeStateEntry *e;
- AudioState *s = &glob_audio_state;
+ AudioState *s;
struct audio_driver *driver;
/* silence gcc warning about uninitialized variable */
AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
- if (s->drv) {
- if (dev) {
- dolog("Cannot create more than one audio backend, sorry\n");
- qapi_free_Audiodev(dev);
- }
- return -1;
- }
-
if (dev) {
/* -audiodev option */
drvname = AudiodevDriver_str(dev->driver);
+ } else if (!QTAILQ_EMPTY(&audio_states)) {
+ /*
+ * todo: check for -audiodev once we have normal audiodev selection
+ * support
+ */
+ return QTAILQ_FIRST(&audio_states);
} else {
/* legacy implicit initialization */
head = audio_handle_legacy_opts();
@@ -1408,12 +1426,18 @@ static int audio_init(Audiodev *dev)
dev = QSIMPLEQ_FIRST(&head)->dev;
audio_validate_opts(dev, &error_abort);
}
+
+ s = g_malloc0(sizeof(AudioState));
s->dev = dev;
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
- atexit(audio_cleanup);
+ if (!atexit_registered) {
+ atexit(audio_cleanup);
+ atexit_registered = true;
+ }
+ QTAILQ_INSERT_TAIL(&audio_states, s, list);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
@@ -1478,7 +1502,7 @@ static int audio_init(Audiodev *dev)
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
- return 0;
+ return s;
}
void audio_free_audiodev_list(AudiodevListHead *head)
@@ -1493,10 +1517,13 @@ void audio_free_audiodev_list(AudiodevListHead *head)
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
- audio_init(NULL);
+ if (!card->state) {
+ card->state = audio_init(NULL);
+ }
+
card->name = g_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
- QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
+ QLIST_INSERT_HEAD (&card->state->card_head, card, entries);
}
void AUD_remove_card (QEMUSoundCard *card)
@@ -1506,16 +1533,21 @@ void AUD_remove_card (QEMUSoundCard *card)
}
-CaptureVoiceOut *AUD_add_capture (
+CaptureVoiceOut *AUD_add_capture(
+ AudioState *s,
struct audsettings *as,
struct audio_capture_ops *ops,
void *cb_opaque
)
{
- AudioState *s = &glob_audio_state;
CaptureVoiceOut *cap;
struct capture_callback *cb;
+ if (!s) {
+ /* todo: remove when we have normal audiodev selection support */
+ s = audio_init(NULL);
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1805,3 +1837,25 @@ int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
return audio_buffer_samples(pdo, as, def_usecs) *
audioformat_bytes_per_sample(as->fmt);
}
+
+AudioState *audio_state_by_name(const char *name)
+{
+ AudioState *s;
+ QTAILQ_FOREACH(s, &audio_states, list) {
+ assert(s->dev);
+ if (strcmp(name, s->dev->id) == 0) {
+ return s;
+ }
+ }
+ return NULL;
+}
+
+const char *audio_get_id(QEMUSoundCard *card)
+{
+ if (card->state) {
+ assert(card->state->dev);
+ return card->state->dev->id;
+ } else {
+ return "";
+ }
+}
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 74320dfecc..81c5c19032 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -105,8 +105,8 @@ static struct capture_ops wav_capture_ops = {
.info = wav_capture_info
};
-int wav_start_capture (CaptureState *s, const char *path, int freq,
- int bits, int nchannels)
+int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
+ int freq, int bits, int nchannels)
{
WAVState *wav;
uint8_t hdr[] = {
@@ -171,7 +171,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
goto error_free;
}
- cap = AUD_add_capture (&as, &ops, wav);
+ cap = AUD_add_capture(state, &as, &ops, wav);
if (!cap) {
error_report("Failed to add audio capture");
goto error_free;
diff --git a/monitor/misc.c b/monitor/misc.c
index 00338c002a..f97810d370 100644
--- a/monitor/misc.c
+++ b/monitor/misc.c
@@ -1148,7 +1148,17 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
int bits = qdict_get_try_int(qdict, "bits", -1);
int has_channels = qdict_haskey(qdict, "nchannels");
int nchannels = qdict_get_try_int(qdict, "nchannels", -1);
+ const char *audiodev = qdict_get_try_str(qdict, "audiodev");
CaptureState *s;
+ AudioState *as = NULL;
+
+ if (audiodev) {
+ as = audio_state_by_name(audiodev);
+ if (!as) {
+ monitor_printf(mon, "Invalid audiodev specified\n");
+ return;
+ }
+ }
s = g_malloc0 (sizeof (*s));
@@ -1156,7 +1166,7 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
bits = has_bits ? bits : 16;
nchannels = has_channels ? nchannels : 2;
- if (wav_start_capture (s, path, freq, bits, nchannels)) {
+ if (wav_start_capture(as, s, path, freq, bits, nchannels)) {
monitor_printf(mon, "Failed to add wave capture\n");
g_free (s);
return;
diff --git a/ui/vnc.c b/ui/vnc.c
index 38f92bfca3..24f9be5b5d 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1222,7 +1222,7 @@ static void audio_add(VncState *vs)
ops.destroy = audio_capture_destroy;
ops.capture = audio_capture;
- vs->audio_cap = AUD_add_capture(&vs->as, &ops, vs);
+ vs->audio_cap = AUD_add_capture(vs->vd->audio_state, &vs->as, &ops, vs);
if (!vs->audio_cap) {
error_report("Failed to add audio capture");
}
@@ -3369,6 +3369,9 @@ static QemuOptsList qemu_vnc_opts = {
},{
.name = "non-adaptive",
.type = QEMU_OPT_BOOL,
+ },{
+ .name = "audiodev",
+ .type = QEMU_OPT_STRING,
},
{ /* end of list */ }
},
@@ -3806,6 +3809,7 @@ void vnc_display_open(const char *id, Error **errp)
const char *saslauthz;
int lock_key_sync = 1;
int key_delay_ms;
+ const char *audiodev;
if (!vd) {
error_setg(errp, "VNC display not active");
@@ -3991,6 +3995,15 @@ void vnc_display_open(const char *id, Error **errp)
}
vd->ledstate = 0;
+ audiodev = qemu_opt_get(opts, "audiodev");
+ if (audiodev) {
+ vd->audio_state = audio_state_by_name(audiodev);
+ if (!vd->audio_state) {
+ error_setg(errp, "Audiodev '%s' not found", audiodev);
+ goto fail;
+ }
+ }
+
device_id = qemu_opt_get(opts, "display");
if (device_id) {
int head = qemu_opt_get_number(opts, "head", 0);
diff --git a/hmp-commands.hx b/hmp-commands.hx
index bfa5681dd2..23196da3fe 100644
--- a/hmp-commands.hx
+++ b/hmp-commands.hx
@@ -819,16 +819,17 @@ ETEXI
{
.name = "wavcapture",
- .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
- .params = "path [frequency [bits [channels]]]",
+ .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
+ .params = "path [frequency [bits [channels [audiodev]]]]",
.help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
.cmd = hmp_wavcapture,
},
STEXI
-@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
+@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
@findex wavcapture
-Capture audio into @var{filename}. Using sample rate @var{frequency}
-bits per sample @var{bits} and number of channels @var{channels}.
+Capture audio into @var{filename} from @var{audiodev}. Using sample rate
+@var{frequency} bits per sample @var{bits} and number of channels
+@var{channels}.
Defaults:
@itemize @minus
diff --git a/qemu-options.hx b/qemu-options.hx
index 9621e934c0..0111055aa4 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
events are arriving in bulk. Possible causes for the latter are flaky
network connections, or scripts for automated testing.
+@item audiodev=@var{audiodev}
+
+Use the specified @var{audiodev} when the VNC client requests audio
+transmission.
+
@end table
ETEXI
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-09 18:35 ` [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio Kővágó, Zoltán
@ 2019-07-10 4:06 ` Markus Armbruster
2019-07-10 19:37 ` Zoltán Kővágó
2019-07-10 19:58 ` Marc-André Lureau
1 sibling, 1 reply; 32+ messages in thread
From: Markus Armbruster @ 2019-07-10 4:06 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: qemu-devel, Dr. David Alan Gilbert, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> Audio functions no longer access glob_audio_state, instead they get an
> AudioState as a parameter. This is required in order to support
> multiple backends.
>
> glob_audio_state is also gone, and replaced with a tailq so we can store
> more than one states.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
[...]
> diff --git a/hmp-commands.hx b/hmp-commands.hx
> index bfa5681dd2..23196da3fe 100644
> --- a/hmp-commands.hx
> +++ b/hmp-commands.hx
> @@ -819,16 +819,17 @@ ETEXI
>
> {
> .name = "wavcapture",
> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
> - .params = "path [frequency [bits [channels]]]",
> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
> + .params = "path [frequency [bits [channels [audiodev]]]]",
> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
> .cmd = hmp_wavcapture,
> },
> STEXI
> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
> @findex wavcapture
> -Capture audio into @var{filename}. Using sample rate @var{frequency}
> -bits per sample @var{bits} and number of channels @var{channels}.
> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
> +@var{frequency} bits per sample @var{bits} and number of channels
> +@var{channels}.
>
> Defaults:
> @itemize @minus
@item Sample rate = 44100 Hz - CD quality
@item Bits = 16
@item Number of channels = 2 - Stereo
@end itemize
ETEXI
Defaults for the other optional arguments are listed here. Why not for
@audiodev?
> diff --git a/qemu-options.hx b/qemu-options.hx
> index 9621e934c0..0111055aa4 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
> events are arriving in bulk. Possible causes for the latter are flaky
> network connections, or scripts for automated testing.
>
> +@item audiodev=@var{audiodev}
> +
> +Use the specified @var{audiodev} when the VNC client requests audio
> +transmission.
> +
What's the default?
> @end table
> ETEXI
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-10 4:06 ` Markus Armbruster
@ 2019-07-10 19:37 ` Zoltán Kővágó
2019-07-11 8:58 ` Dr. David Alan Gilbert
2019-07-11 14:37 ` Markus Armbruster
0 siblings, 2 replies; 32+ messages in thread
From: Zoltán Kővágó @ 2019-07-10 19:37 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Dr. David Alan Gilbert, Gerd Hoffmann
On 2019-07-10 06:06, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> Audio functions no longer access glob_audio_state, instead they get an
>> AudioState as a parameter. This is required in order to support
>> multiple backends.
>>
>> glob_audio_state is also gone, and replaced with a tailq so we can store
>> more than one states.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
> [...]
>> diff --git a/hmp-commands.hx b/hmp-commands.hx
>> index bfa5681dd2..23196da3fe 100644
>> --- a/hmp-commands.hx
>> +++ b/hmp-commands.hx
>> @@ -819,16 +819,17 @@ ETEXI
>>
>> {
>> .name = "wavcapture",
>> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
>> - .params = "path [frequency [bits [channels]]]",
>> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
>> + .params = "path [frequency [bits [channels [audiodev]]]]",
>> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
>> .cmd = hmp_wavcapture,
>> },
>> STEXI
>> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
>> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
>> @findex wavcapture
>> -Capture audio into @var{filename}. Using sample rate @var{frequency}
>> -bits per sample @var{bits} and number of channels @var{channels}.
>> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
>> +@var{frequency} bits per sample @var{bits} and number of channels
>> +@var{channels}.
>>
>> Defaults:
>> @itemize @minus
> @item Sample rate = 44100 Hz - CD quality
> @item Bits = 16
> @item Number of channels = 2 - Stereo
> @end itemize
> ETEXI
>
> Defaults for the other optional arguments are listed here. Why not for
> @audiodev?
There's no default listed because there's no default when you use the
-audiodev options, since there's no good default. When you don't use
-audiodev, it'll use the implicitly created audiodev which doesn't have
a name, so it can't be specified. But I agree that this situation
should be documented somehow.
>
>> diff --git a/qemu-options.hx b/qemu-options.hx
>> index 9621e934c0..0111055aa4 100644
>> --- a/qemu-options.hx
>> +++ b/qemu-options.hx
>> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
>> events are arriving in bulk. Possible causes for the latter are flaky
>> network connections, or scripts for automated testing.
>>
>> +@item audiodev=@var{audiodev}
>> +
>> +Use the specified @var{audiodev} when the VNC client requests audio
>> +transmission.
>> +
>
> What's the default?
It's the same story as wav_capture.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-10 19:37 ` Zoltán Kővágó
@ 2019-07-11 8:58 ` Dr. David Alan Gilbert
2019-07-11 14:37 ` Markus Armbruster
1 sibling, 0 replies; 32+ messages in thread
From: Dr. David Alan Gilbert @ 2019-07-11 8:58 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Gerd Hoffmann, Markus Armbruster, qemu-devel
* Zoltán Kővágó (dirty.ice.hu@gmail.com) wrote:
> On 2019-07-10 06:06, Markus Armbruster wrote:
> > "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> >
> >> Audio functions no longer access glob_audio_state, instead they get an
> >> AudioState as a parameter. This is required in order to support
> >> multiple backends.
> >>
> >> glob_audio_state is also gone, and replaced with a tailq so we can store
> >> more than one states.
> >>
> >> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> >> ---
> > [...]
> >> diff --git a/hmp-commands.hx b/hmp-commands.hx
> >> index bfa5681dd2..23196da3fe 100644
> >> --- a/hmp-commands.hx
> >> +++ b/hmp-commands.hx
> >> @@ -819,16 +819,17 @@ ETEXI
> >>
> >> {
> >> .name = "wavcapture",
> >> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
> >> - .params = "path [frequency [bits [channels]]]",
> >> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
> >> + .params = "path [frequency [bits [channels [audiodev]]]]",
> >> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
> >> .cmd = hmp_wavcapture,
> >> },
> >> STEXI
> >> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
> >> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
> >> @findex wavcapture
> >> -Capture audio into @var{filename}. Using sample rate @var{frequency}
> >> -bits per sample @var{bits} and number of channels @var{channels}.
> >> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
> >> +@var{frequency} bits per sample @var{bits} and number of channels
> >> +@var{channels}.
> >>
> >> Defaults:
> >> @itemize @minus
> > @item Sample rate = 44100 Hz - CD quality
> > @item Bits = 16
> > @item Number of channels = 2 - Stereo
> > @end itemize
> > ETEXI
> >
> > Defaults for the other optional arguments are listed here. Why not for
> > @audiodev?
>
> There's no default listed because there's no default when you use the
> -audiodev options, since there's no good default. When you don't use
> -audiodev, it'll use the implicitly created audiodev which doesn't have
> a name, so it can't be specified. But I agree that this situation
> should be documented somehow.
>
> >
> >> diff --git a/qemu-options.hx b/qemu-options.hx
> >> index 9621e934c0..0111055aa4 100644
> >> --- a/qemu-options.hx
> >> +++ b/qemu-options.hx
> >> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
> >> events are arriving in bulk. Possible causes for the latter are flaky
> >> network connections, or scripts for automated testing.
> >>
> >> +@item audiodev=@var{audiodev}
> >> +
> >> +Use the specified @var{audiodev} when the VNC client requests audio
> >> +transmission.
> >> +
> >
> > What's the default?
>
> It's the same story as wav_capture.
OK, so I think you're saying the default behaviour doesn't change.
Add a comment to document that, and that seems OK.
Dave
> Regards,
> Zoltan
--
Dr. David Alan Gilbert / dgilbert@redhat.com / Manchester, UK
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-10 19:37 ` Zoltán Kővágó
2019-07-11 8:58 ` Dr. David Alan Gilbert
@ 2019-07-11 14:37 ` Markus Armbruster
2019-07-11 19:07 ` Zoltán Kővágó
1 sibling, 1 reply; 32+ messages in thread
From: Markus Armbruster @ 2019-07-11 14:37 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Gerd Hoffmann, qemu-devel, Dr. David Alan Gilbert
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-07-10 06:06, Markus Armbruster wrote:
>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>
>>> Audio functions no longer access glob_audio_state, instead they get an
>>> AudioState as a parameter. This is required in order to support
>>> multiple backends.
>>>
>>> glob_audio_state is also gone, and replaced with a tailq so we can store
>>> more than one states.
>>>
>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>> ---
>> [...]
>>> diff --git a/hmp-commands.hx b/hmp-commands.hx
>>> index bfa5681dd2..23196da3fe 100644
>>> --- a/hmp-commands.hx
>>> +++ b/hmp-commands.hx
>>> @@ -819,16 +819,17 @@ ETEXI
>>>
>>> {
>>> .name = "wavcapture",
>>> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
>>> - .params = "path [frequency [bits [channels]]]",
>>> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
>>> + .params = "path [frequency [bits [channels [audiodev]]]]",
>>> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
>>> .cmd = hmp_wavcapture,
>>> },
>>> STEXI
>>> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
>>> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
>>> @findex wavcapture
>>> -Capture audio into @var{filename}. Using sample rate @var{frequency}
>>> -bits per sample @var{bits} and number of channels @var{channels}.
>>> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
>>> +@var{frequency} bits per sample @var{bits} and number of channels
>>> +@var{channels}.
>>>
>>> Defaults:
>>> @itemize @minus
>> @item Sample rate = 44100 Hz - CD quality
>> @item Bits = 16
>> @item Number of channels = 2 - Stereo
>> @end itemize
>> ETEXI
>>
>> Defaults for the other optional arguments are listed here. Why not for
>> @audiodev?
>
> There's no default listed because there's no default when you use the
> -audiodev options, since there's no good default. When you don't use
> -audiodev, it'll use the implicitly created audiodev which doesn't have
> a name, so it can't be specified.
Double-checking to avoid misunderstandings: there is a default
*behavior*, but no default *value*, i.e. there is no VALUE that makes
audiodev=VALUE give you the same behavior as no audiodev. Correct?
> But I agree that this situation
> should be documented somehow.
Yes, please.
>>> diff --git a/qemu-options.hx b/qemu-options.hx
>>> index 9621e934c0..0111055aa4 100644
>>> --- a/qemu-options.hx
>>> +++ b/qemu-options.hx
>>> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
>>> events are arriving in bulk. Possible causes for the latter are flaky
>>> network connections, or scripts for automated testing.
>>>
>>> +@item audiodev=@var{audiodev}
>>> +
>>> +Use the specified @var{audiodev} when the VNC client requests audio
>>> +transmission.
>>> +
>>
>> What's the default?
>
> It's the same story as wav_capture.
>
> Regards,
> Zoltan
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-11 14:37 ` Markus Armbruster
@ 2019-07-11 19:07 ` Zoltán Kővágó
0 siblings, 0 replies; 32+ messages in thread
From: Zoltán Kővágó @ 2019-07-11 19:07 UTC (permalink / raw)
To: Markus Armbruster; +Cc: Gerd Hoffmann, qemu-devel, Dr. David Alan Gilbert
On 2019-07-11 16:37, Markus Armbruster wrote:
> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>
>> On 2019-07-10 06:06, Markus Armbruster wrote:
>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>
>>>> Audio functions no longer access glob_audio_state, instead they get an
>>>> AudioState as a parameter. This is required in order to support
>>>> multiple backends.
>>>>
>>>> glob_audio_state is also gone, and replaced with a tailq so we can store
>>>> more than one states.
>>>>
>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>> ---
>>> [...]
>>>> diff --git a/hmp-commands.hx b/hmp-commands.hx
>>>> index bfa5681dd2..23196da3fe 100644
>>>> --- a/hmp-commands.hx
>>>> +++ b/hmp-commands.hx
>>>> @@ -819,16 +819,17 @@ ETEXI
>>>>
>>>> {
>>>> .name = "wavcapture",
>>>> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
>>>> - .params = "path [frequency [bits [channels]]]",
>>>> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
>>>> + .params = "path [frequency [bits [channels [audiodev]]]]",
>>>> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
>>>> .cmd = hmp_wavcapture,
>>>> },
>>>> STEXI
>>>> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
>>>> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
>>>> @findex wavcapture
>>>> -Capture audio into @var{filename}. Using sample rate @var{frequency}
>>>> -bits per sample @var{bits} and number of channels @var{channels}.
>>>> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
>>>> +@var{frequency} bits per sample @var{bits} and number of channels
>>>> +@var{channels}.
>>>>
>>>> Defaults:
>>>> @itemize @minus
>>> @item Sample rate = 44100 Hz - CD quality
>>> @item Bits = 16
>>> @item Number of channels = 2 - Stereo
>>> @end itemize
>>> ETEXI
>>>
>>> Defaults for the other optional arguments are listed here. Why not for
>>> @audiodev?
>>
>> There's no default listed because there's no default when you use the
>> -audiodev options, since there's no good default. When you don't use
>> -audiodev, it'll use the implicitly created audiodev which doesn't have
>> a name, so it can't be specified.
>
> Double-checking to avoid misunderstandings: there is a default
> *behavior*, but no default *value*, i.e. there is no VALUE that makes
> audiodev=VALUE give you the same behavior as no audiodev. Correct?
Yes. If there is no audiodev=VALUE, and no -audiodev on the command
line, use the legacy config. If there is audiodev=VALUE and -audiodev
id=VALUE, use that device. Otherwise, it's an error.
>
>> But I agree that this situation
>> should be documented somehow.
>
> Yes, please.
>
>>>> diff --git a/qemu-options.hx b/qemu-options.hx
>>>> index 9621e934c0..0111055aa4 100644
>>>> --- a/qemu-options.hx
>>>> +++ b/qemu-options.hx
>>>> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
>>>> events are arriving in bulk. Possible causes for the latter are flaky
>>>> network connections, or scripts for automated testing.
>>>>
>>>> +@item audiodev=@var{audiodev}
>>>> +
>>>> +Use the specified @var{audiodev} when the VNC client requests audio
>>>> +transmission.
>>>> +
>>>
>>> What's the default?
>>
>> It's the same story as wav_capture.
>>
>> Regards,
>> Zoltan
^ permalink raw reply [flat|nested] 32+ messages in thread
* Re: [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio
2019-07-09 18:35 ` [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio Kővágó, Zoltán
2019-07-10 4:06 ` Markus Armbruster
@ 2019-07-10 19:58 ` Marc-André Lureau
1 sibling, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: Gerd Hoffmann, QEMU, Dr. David Alan Gilbert, Markus Armbruster
On Tue, Jul 9, 2019 at 11:02 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Audio functions no longer access glob_audio_state, instead they get an
> AudioState as a parameter. This is required in order to support
> multiple backends.
>
> glob_audio_state is also gone, and replaced with a tailq so we can store
> more than one states.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
I guess you could have made the vnc and wavcapture audiodev argument a
seperate commit. Or mention it in commit message.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> audio/audio.h | 12 +++--
> audio/audio_int.h | 2 +
> audio/audio_template.h | 2 +-
> ui/vnc.h | 2 +
> audio/audio.c | 102 +++++++++++++++++++++++++++++++----------
> audio/wavcapture.c | 6 +--
> monitor/misc.c | 12 ++++-
> ui/vnc.c | 15 +++++-
> hmp-commands.hx | 11 +++--
> qemu-options.hx | 5 ++
> 10 files changed, 131 insertions(+), 38 deletions(-)
>
> diff --git a/audio/audio.h b/audio/audio.h
> index 64b0f761bc..ad2457f4de 100644
> --- a/audio/audio.h
> +++ b/audio/audio.h
> @@ -78,8 +78,10 @@ typedef struct SWVoiceOut SWVoiceOut;
> typedef struct CaptureVoiceOut CaptureVoiceOut;
> typedef struct SWVoiceIn SWVoiceIn;
>
> +typedef struct AudioState AudioState;
> typedef struct QEMUSoundCard {
> char *name;
> + AudioState *state;
> QLIST_ENTRY (QEMUSoundCard) entries;
> } QEMUSoundCard;
>
> @@ -92,7 +94,8 @@ void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
>
> void AUD_register_card (const char *name, QEMUSoundCard *card);
> void AUD_remove_card (QEMUSoundCard *card);
> -CaptureVoiceOut *AUD_add_capture (
> +CaptureVoiceOut *AUD_add_capture(
> + AudioState *s,
> struct audsettings *as,
> struct audio_capture_ops *ops,
> void *opaque
> @@ -160,8 +163,8 @@ static inline void *advance (void *p, int incr)
> #define audio_MAX(a, b) ((a)<(b)?(b):(a))
> #endif
>
> -int wav_start_capture (CaptureState *s, const char *path, int freq,
> - int bits, int nchannels);
> +int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
> + int freq, int bits, int nchannels);
>
> bool audio_is_cleaning_up(void);
> void audio_cleanup(void);
> @@ -175,4 +178,7 @@ void audio_parse_option(const char *opt);
> void audio_init_audiodevs(void);
> void audio_legacy_help(void);
>
> +AudioState *audio_state_by_name(const char *name);
> +const char *audio_get_id(QEMUSoundCard *card);
> +
> #endif /* QEMU_AUDIO_H */
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index 8164696b2c..9f01f6ad00 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -196,6 +196,8 @@ typedef struct AudioState {
>
> bool timer_running;
> uint64_t timer_last;
> +
> + QTAILQ_ENTRY(AudioState) list;
> } AudioState;
>
> extern const struct mixeng_volume nominal_volume;
> diff --git a/audio/audio_template.h b/audio/audio_template.h
> index c721fed75d..54f07338e7 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -428,7 +428,7 @@ SW *glue (AUD_open_, TYPE) (
> struct audsettings *as
> )
> {
> - AudioState *s = &glob_audio_state;
> + AudioState *s = card->state;
> AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
>
> if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
> diff --git a/ui/vnc.h b/ui/vnc.h
> index 2f84db3142..6f54653455 100644
> --- a/ui/vnc.h
> +++ b/ui/vnc.h
> @@ -183,6 +183,8 @@ struct VncDisplay
> #ifdef CONFIG_VNC_SASL
> VncDisplaySASL sasl;
> #endif
> +
> + AudioState *audio_state;
> };
>
> typedef struct VncTight {
> diff --git a/audio/audio.c b/audio/audio.c
> index 8d2f580788..e9dd7c8b32 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -87,7 +87,8 @@ audio_driver *audio_driver_lookup(const char *name)
> return NULL;
> }
>
> -static AudioState glob_audio_state;
> +static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
> + QTAILQ_HEAD_INITIALIZER(audio_states);
>
> const struct mixeng_volume nominal_volume = {
> .mute = 0,
> @@ -1236,11 +1237,14 @@ static void audio_run_capture (AudioState *s)
>
> void audio_run (const char *msg)
> {
> - AudioState *s = &glob_audio_state;
> + AudioState *s;
> +
> + QTAILQ_FOREACH(s, &audio_states, list) {
> + audio_run_out(s);
> + audio_run_in(s);
> + audio_run_capture(s);
> + }
>
> - audio_run_out (s);
> - audio_run_in (s);
> - audio_run_capture (s);
> #ifdef DEBUG_POLL
> {
> static double prevtime;
> @@ -1304,13 +1308,11 @@ bool audio_is_cleaning_up(void)
> return is_cleaning_up;
> }
>
> -void audio_cleanup(void)
> +static void free_audio_state(AudioState *s)
> {
> - AudioState *s = &glob_audio_state;
> HWVoiceOut *hwo, *hwon;
> HWVoiceIn *hwi, *hwin;
>
> - is_cleaning_up = true;
> QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
> SWVoiceCap *sc;
>
> @@ -1347,6 +1349,17 @@ void audio_cleanup(void)
> qapi_free_Audiodev(s->dev);
> s->dev = NULL;
> }
> + g_free(s);
> +}
> +
> +void audio_cleanup(void)
> +{
> + is_cleaning_up = true;
> + while (!QTAILQ_EMPTY(&audio_states)) {
> + AudioState *s = QTAILQ_FIRST(&audio_states);
> + QTAILQ_REMOVE(&audio_states, s, list);
> + free_audio_state(s);
> + }
> }
>
> static const VMStateDescription vmstate_audio = {
> @@ -1373,28 +1386,33 @@ static AudiodevListEntry *audiodev_find(
> return NULL;
> }
>
> -static int audio_init(Audiodev *dev)
> +/*
> + * if we have dev, this function was called because of an -audiodev argument =>
> + * initialize a new state with it
> + * if dev == NULL => legacy implicit initialization, return the already created
> + * state or create a new one
> + */
> +static AudioState *audio_init(Audiodev *dev)
> {
> + static bool atexit_registered;
> size_t i;
> int done = 0;
> const char *drvname = NULL;
> VMChangeStateEntry *e;
> - AudioState *s = &glob_audio_state;
> + AudioState *s;
> struct audio_driver *driver;
> /* silence gcc warning about uninitialized variable */
> AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
>
> - if (s->drv) {
> - if (dev) {
> - dolog("Cannot create more than one audio backend, sorry\n");
> - qapi_free_Audiodev(dev);
> - }
> - return -1;
> - }
> -
> if (dev) {
> /* -audiodev option */
> drvname = AudiodevDriver_str(dev->driver);
> + } else if (!QTAILQ_EMPTY(&audio_states)) {
> + /*
> + * todo: check for -audiodev once we have normal audiodev selection
> + * support
> + */
> + return QTAILQ_FIRST(&audio_states);
> } else {
> /* legacy implicit initialization */
> head = audio_handle_legacy_opts();
> @@ -1408,12 +1426,18 @@ static int audio_init(Audiodev *dev)
> dev = QSIMPLEQ_FIRST(&head)->dev;
> audio_validate_opts(dev, &error_abort);
> }
> +
> + s = g_malloc0(sizeof(AudioState));
> s->dev = dev;
>
> QLIST_INIT (&s->hw_head_out);
> QLIST_INIT (&s->hw_head_in);
> QLIST_INIT (&s->cap_head);
> - atexit(audio_cleanup);
> + if (!atexit_registered) {
> + atexit(audio_cleanup);
> + atexit_registered = true;
> + }
> + QTAILQ_INSERT_TAIL(&audio_states, s, list);
>
> s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
>
> @@ -1478,7 +1502,7 @@ static int audio_init(Audiodev *dev)
>
> QLIST_INIT (&s->card_head);
> vmstate_register (NULL, 0, &vmstate_audio, s);
> - return 0;
> + return s;
> }
>
> void audio_free_audiodev_list(AudiodevListHead *head)
> @@ -1493,10 +1517,13 @@ void audio_free_audiodev_list(AudiodevListHead *head)
>
> void AUD_register_card (const char *name, QEMUSoundCard *card)
> {
> - audio_init(NULL);
> + if (!card->state) {
> + card->state = audio_init(NULL);
> + }
> +
> card->name = g_strdup (name);
> memset (&card->entries, 0, sizeof (card->entries));
> - QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
> + QLIST_INSERT_HEAD (&card->state->card_head, card, entries);
> }
>
> void AUD_remove_card (QEMUSoundCard *card)
> @@ -1506,16 +1533,21 @@ void AUD_remove_card (QEMUSoundCard *card)
> }
>
>
> -CaptureVoiceOut *AUD_add_capture (
> +CaptureVoiceOut *AUD_add_capture(
> + AudioState *s,
> struct audsettings *as,
> struct audio_capture_ops *ops,
> void *cb_opaque
> )
> {
> - AudioState *s = &glob_audio_state;
> CaptureVoiceOut *cap;
> struct capture_callback *cb;
>
> + if (!s) {
> + /* todo: remove when we have normal audiodev selection support */
> + s = audio_init(NULL);
> + }
> +
> if (audio_validate_settings (as)) {
> dolog ("Invalid settings were passed when trying to add capture\n");
> audio_print_settings (as);
> @@ -1805,3 +1837,25 @@ int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
> return audio_buffer_samples(pdo, as, def_usecs) *
> audioformat_bytes_per_sample(as->fmt);
> }
> +
> +AudioState *audio_state_by_name(const char *name)
> +{
> + AudioState *s;
> + QTAILQ_FOREACH(s, &audio_states, list) {
> + assert(s->dev);
> + if (strcmp(name, s->dev->id) == 0) {
> + return s;
> + }
> + }
> + return NULL;
> +}
> +
> +const char *audio_get_id(QEMUSoundCard *card)
> +{
> + if (card->state) {
> + assert(card->state->dev);
> + return card->state->dev->id;
> + } else {
> + return "";
> + }
> +}
> diff --git a/audio/wavcapture.c b/audio/wavcapture.c
> index 74320dfecc..81c5c19032 100644
> --- a/audio/wavcapture.c
> +++ b/audio/wavcapture.c
> @@ -105,8 +105,8 @@ static struct capture_ops wav_capture_ops = {
> .info = wav_capture_info
> };
>
> -int wav_start_capture (CaptureState *s, const char *path, int freq,
> - int bits, int nchannels)
> +int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
> + int freq, int bits, int nchannels)
> {
> WAVState *wav;
> uint8_t hdr[] = {
> @@ -171,7 +171,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
> goto error_free;
> }
>
> - cap = AUD_add_capture (&as, &ops, wav);
> + cap = AUD_add_capture(state, &as, &ops, wav);
> if (!cap) {
> error_report("Failed to add audio capture");
> goto error_free;
> diff --git a/monitor/misc.c b/monitor/misc.c
> index 00338c002a..f97810d370 100644
> --- a/monitor/misc.c
> +++ b/monitor/misc.c
> @@ -1148,7 +1148,17 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
> int bits = qdict_get_try_int(qdict, "bits", -1);
> int has_channels = qdict_haskey(qdict, "nchannels");
> int nchannels = qdict_get_try_int(qdict, "nchannels", -1);
> + const char *audiodev = qdict_get_try_str(qdict, "audiodev");
> CaptureState *s;
> + AudioState *as = NULL;
> +
> + if (audiodev) {
> + as = audio_state_by_name(audiodev);
> + if (!as) {
> + monitor_printf(mon, "Invalid audiodev specified\n");
> + return;
> + }
> + }
>
> s = g_malloc0 (sizeof (*s));
>
> @@ -1156,7 +1166,7 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
> bits = has_bits ? bits : 16;
> nchannels = has_channels ? nchannels : 2;
>
> - if (wav_start_capture (s, path, freq, bits, nchannels)) {
> + if (wav_start_capture(as, s, path, freq, bits, nchannels)) {
> monitor_printf(mon, "Failed to add wave capture\n");
> g_free (s);
> return;
> diff --git a/ui/vnc.c b/ui/vnc.c
> index 38f92bfca3..24f9be5b5d 100644
> --- a/ui/vnc.c
> +++ b/ui/vnc.c
> @@ -1222,7 +1222,7 @@ static void audio_add(VncState *vs)
> ops.destroy = audio_capture_destroy;
> ops.capture = audio_capture;
>
> - vs->audio_cap = AUD_add_capture(&vs->as, &ops, vs);
> + vs->audio_cap = AUD_add_capture(vs->vd->audio_state, &vs->as, &ops, vs);
> if (!vs->audio_cap) {
> error_report("Failed to add audio capture");
> }
> @@ -3369,6 +3369,9 @@ static QemuOptsList qemu_vnc_opts = {
> },{
> .name = "non-adaptive",
> .type = QEMU_OPT_BOOL,
> + },{
> + .name = "audiodev",
> + .type = QEMU_OPT_STRING,
> },
> { /* end of list */ }
> },
> @@ -3806,6 +3809,7 @@ void vnc_display_open(const char *id, Error **errp)
> const char *saslauthz;
> int lock_key_sync = 1;
> int key_delay_ms;
> + const char *audiodev;
>
> if (!vd) {
> error_setg(errp, "VNC display not active");
> @@ -3991,6 +3995,15 @@ void vnc_display_open(const char *id, Error **errp)
> }
> vd->ledstate = 0;
>
> + audiodev = qemu_opt_get(opts, "audiodev");
> + if (audiodev) {
> + vd->audio_state = audio_state_by_name(audiodev);
> + if (!vd->audio_state) {
> + error_setg(errp, "Audiodev '%s' not found", audiodev);
> + goto fail;
> + }
> + }
> +
> device_id = qemu_opt_get(opts, "display");
> if (device_id) {
> int head = qemu_opt_get_number(opts, "head", 0);
> diff --git a/hmp-commands.hx b/hmp-commands.hx
> index bfa5681dd2..23196da3fe 100644
> --- a/hmp-commands.hx
> +++ b/hmp-commands.hx
> @@ -819,16 +819,17 @@ ETEXI
>
> {
> .name = "wavcapture",
> - .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
> - .params = "path [frequency [bits [channels]]]",
> + .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
> + .params = "path [frequency [bits [channels [audiodev]]]]",
> .help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
> .cmd = hmp_wavcapture,
> },
> STEXI
> -@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
> +@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
> @findex wavcapture
> -Capture audio into @var{filename}. Using sample rate @var{frequency}
> -bits per sample @var{bits} and number of channels @var{channels}.
> +Capture audio into @var{filename} from @var{audiodev}. Using sample rate
> +@var{frequency} bits per sample @var{bits} and number of channels
> +@var{channels}.
>
> Defaults:
> @itemize @minus
> diff --git a/qemu-options.hx b/qemu-options.hx
> index 9621e934c0..0111055aa4 100644
> --- a/qemu-options.hx
> +++ b/qemu-options.hx
> @@ -1978,6 +1978,11 @@ can help the device and guest to keep up and not lose events in case
> events are arriving in bulk. Possible causes for the latter are flaky
> network connections, or scripts for automated testing.
>
> +@item audiodev=@var{audiodev}
> +
> +Use the specified @var{audiodev} when the VNC client requests audio
> +transmission.
> +
> @end table
> ETEXI
>
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 03/11] audio: add audiodev properties to frontends
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
2019-07-09 18:35 ` [Qemu-devel] [PATCH 01/11] audio: reduce glob_audio_state usage Kővágó, Zoltán
2019-07-09 18:35 ` [Qemu-devel] [PATCH 02/11] audio: basic support for multi backend audio Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
` (9 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel
Cc: Paolo Bonzini, Michael Walle, Daniel P. Berrangé,
Gerd Hoffmann, Eduardo Habkost
Finally add audiodev= options to audio frontends so users can specify
which backend to use when multiple backends exist. Not specifying an
audiodev= option currently causes the first audiodev to be used, this is
fixed in the next commit.
Example usage: -audiodev pa,id=foo -device AC97,audiodev=foo
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.h | 3 ++
include/hw/qdev-properties.h | 3 ++
hw/audio/ac97.c | 1 +
hw/audio/adlib.c | 1 +
hw/audio/cs4231a.c | 1 +
hw/audio/es1370.c | 7 +++-
hw/audio/gus.c | 1 +
hw/audio/hda-codec.c | 1 +
hw/audio/milkymist-ac97.c | 6 ++++
hw/audio/pcspk.c | 1 +
hw/audio/pl041.c | 1 +
hw/audio/sb16.c | 1 +
hw/audio/wm8750.c | 6 ++++
hw/core/qdev-properties-system.c | 57 ++++++++++++++++++++++++++++++++
hw/usb/dev-audio.c | 1 +
15 files changed, 90 insertions(+), 1 deletion(-)
diff --git a/audio/audio.h b/audio/audio.h
index ad2457f4de..c0722a5cda 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -181,4 +181,7 @@ void audio_legacy_help(void);
AudioState *audio_state_by_name(const char *name);
const char *audio_get_id(QEMUSoundCard *card);
+#define DEFINE_AUDIO_PROPERTIES(_s, _f) \
+ DEFINE_PROP_AUDIODEV("audiodev", _s, _f)
+
#endif /* QEMU_AUDIO_H */
diff --git a/include/hw/qdev-properties.h b/include/hw/qdev-properties.h
index 1eae5ab056..de0b2c8423 100644
--- a/include/hw/qdev-properties.h
+++ b/include/hw/qdev-properties.h
@@ -35,6 +35,7 @@ extern const PropertyInfo qdev_prop_blocksize;
extern const PropertyInfo qdev_prop_pci_host_devaddr;
extern const PropertyInfo qdev_prop_uuid;
extern const PropertyInfo qdev_prop_arraylen;
+extern const PropertyInfo qdev_prop_audiodev;
extern const PropertyInfo qdev_prop_link;
extern const PropertyInfo qdev_prop_off_auto_pcibar;
extern const PropertyInfo qdev_prop_pcie_link_speed;
@@ -236,6 +237,8 @@ extern const PropertyInfo qdev_prop_pcie_link_width;
+ type_check(QemuUUID, typeof_field(_state, _field)), \
.set_default = true, \
}
+#define DEFINE_PROP_AUDIODEV(_n, _s, _f) \
+ DEFINE_PROP(_n, _s, _f, qdev_prop_audiodev, QEMUSoundCard)
#define DEFINE_PROP_END_OF_LIST() \
{}
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index fb98da2678..0d8e524233 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -1409,6 +1409,7 @@ static int ac97_init (PCIBus *bus)
}
static Property ac97_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(AC97LinkState, card),
DEFINE_PROP_UINT32 ("use_broken_id", AC97LinkState, use_broken_id, 0),
DEFINE_PROP_END_OF_LIST (),
};
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 7dd9a89b89..df2e781788 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -299,6 +299,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
}
static Property adlib_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(AdlibState, card),
DEFINE_PROP_UINT32 ("iobase", AdlibState, port, 0x220),
DEFINE_PROP_UINT32 ("freq", AdlibState, freq, 44100),
DEFINE_PROP_END_OF_LIST (),
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 7216b41cc1..e3ea830b47 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -689,6 +689,7 @@ static int cs4231a_init (ISABus *bus)
}
static Property cs4231a_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(CSState, card),
DEFINE_PROP_UINT32 ("iobase", CSState, port, 0x534),
DEFINE_PROP_UINT32 ("irq", CSState, irq, 9),
DEFINE_PROP_UINT32 ("dma", CSState, dma, 3),
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 260c142b70..7589671d20 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -887,6 +887,11 @@ static int es1370_init (PCIBus *bus)
return 0;
}
+static Property es1370_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(ES1370State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void es1370_class_init (ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS (klass);
@@ -903,6 +908,7 @@ static void es1370_class_init (ObjectClass *klass, void *data)
dc->desc = "ENSONIQ AudioPCI ES1370";
dc->vmsd = &vmstate_es1370;
dc->reset = es1370_on_reset;
+ dc->props = es1370_properties;
}
static const TypeInfo es1370_info = {
@@ -923,4 +929,3 @@ static void es1370_register_types (void)
}
type_init (es1370_register_types)
-
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 9ab51631d9..566864bc9e 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -297,6 +297,7 @@ static int GUS_init (ISABus *bus)
}
static Property gus_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(GUSState, card),
DEFINE_PROP_UINT32 ("freq", GUSState, freq, 44100),
DEFINE_PROP_UINT32 ("iobase", GUSState, port, 0x240),
DEFINE_PROP_UINT32 ("irq", GUSState, emu.gusirq, 7),
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 84639b5424..967a10f189 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -840,6 +840,7 @@ static const VMStateDescription vmstate_hda_audio = {
};
static Property hda_audio_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
DEFINE_PROP_UINT32("debug", HDAAudioState, debug, 0),
DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true),
DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer, true),
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index bf6a5a6b96..4835229326 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -329,6 +329,11 @@ static const VMStateDescription vmstate_milkymist_ac97 = {
}
};
+static Property milkymist_ac97_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(MilkymistAC97State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
@@ -336,6 +341,7 @@ static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
dc->realize = milkymist_ac97_realize;
dc->reset = milkymist_ac97_reset;
dc->vmsd = &vmstate_milkymist_ac97;
+ dc->props = milkymist_ac97_properties;
}
static const TypeInfo milkymist_ac97_info = {
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index a7b3ab0c2b..01127304c2 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -209,6 +209,7 @@ static const VMStateDescription vmstate_spk = {
};
static Property pcspk_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(PCSpkState, card),
DEFINE_PROP_UINT32("iobase", PCSpkState, iobase, -1),
DEFINE_PROP_BOOL("migrate", PCSpkState, migrate, true),
DEFINE_PROP_END_OF_LIST(),
diff --git a/hw/audio/pl041.c b/hw/audio/pl041.c
index 59c6ce1f2e..ce430a7821 100644
--- a/hw/audio/pl041.c
+++ b/hw/audio/pl041.c
@@ -622,6 +622,7 @@ static const VMStateDescription vmstate_pl041 = {
};
static Property pl041_device_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(PL041State, codec.card),
/* Non-compact FIFO depth property */
DEFINE_PROP_UINT32("nc_fifo_depth", PL041State, fifo_depth,
DEFAULT_FIFO_DEPTH),
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index a2db06a2bb..6b604979cf 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1421,6 +1421,7 @@ static int SB16_init (ISABus *bus)
}
static Property sb16_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(SB16State, card),
DEFINE_PROP_UINT32 ("version", SB16State, ver, 0x0405), /* 4.5 */
DEFINE_PROP_UINT32 ("iobase", SB16State, port, 0x220),
DEFINE_PROP_UINT32 ("irq", SB16State, irq, 5),
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 81f83eea88..dfb4156ff4 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -701,6 +701,11 @@ void wm8750_set_bclk_in(void *opaque, int new_hz)
wm8750_clk_update(s, 1);
}
+static Property wm8750_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(WM8750State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void wm8750_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
@@ -711,6 +716,7 @@ static void wm8750_class_init(ObjectClass *klass, void *data)
sc->recv = wm8750_rx;
sc->send = wm8750_tx;
dc->vmsd = &vmstate_wm8750;
+ dc->props = wm8750_properties;
}
static const TypeInfo wm8750_info = {
diff --git a/hw/core/qdev-properties-system.c b/hw/core/qdev-properties-system.c
index ba412dd2ca..bf9974370c 100644
--- a/hw/core/qdev-properties-system.c
+++ b/hw/core/qdev-properties-system.c
@@ -11,6 +11,7 @@
*/
#include "qemu/osdep.h"
+#include "audio/audio.h"
#include "net/net.h"
#include "hw/qdev.h"
#include "qapi/error.h"
@@ -353,6 +354,62 @@ const PropertyInfo qdev_prop_netdev = {
};
+/* --- audiodev --- */
+static void get_audiodev(Object *obj, Visitor *v, const char* name,
+ void *opaque, Error **errp)
+{
+ DeviceState *dev = DEVICE(obj);
+ Property *prop = opaque;
+ QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
+ char *p = g_strdup(audio_get_id(card));
+
+ visit_type_str(v, name, &p, errp);
+ g_free(p);
+}
+
+static void set_audiodev(Object *obj, Visitor *v, const char* name,
+ void *opaque, Error **errp)
+{
+ DeviceState *dev = DEVICE(obj);
+ Property *prop = opaque;
+ QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
+ AudioState *state;
+ Error *local_err = NULL;
+ int err = 0;
+ char *str;
+
+ if (dev->realized) {
+ qdev_prop_set_after_realize(dev, name, errp);
+ return;
+ }
+
+ visit_type_str(v, name, &str, &local_err);
+ if (local_err) {
+ error_propagate(errp, local_err);
+ return;
+ }
+
+ state = audio_state_by_name(str);
+
+ if (!state) {
+ err = -ENOENT;
+ goto out;
+ }
+ card->state = state;
+
+out:
+ error_set_from_qdev_prop_error(errp, err, dev, prop, str);
+ g_free(str);
+}
+
+const PropertyInfo qdev_prop_audiodev = {
+ .name = "str",
+ .description = "ID of an audiodev to use as a backend",
+ /* release done on shutdown */
+ .get = get_audiodev,
+ .set = set_audiodev,
+};
+
void qdev_prop_set_drive(DeviceState *dev, const char *name,
BlockBackend *value, Error **errp)
{
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 718ab53cbb..1b1cf91132 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -666,6 +666,7 @@ static const VMStateDescription vmstate_usb_audio = {
};
static Property usb_audio_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(USBAudioState, card),
DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0),
DEFINE_PROP_UINT32("buffer", USBAudioState, buffer,
32 * USBAUDIO_PACKET_SIZE),
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 03/11] audio: add audiodev properties to frontends
2019-07-09 18:35 ` [Qemu-devel] [PATCH 03/11] audio: add audiodev properties to frontends Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: Daniel P. Berrangé,
Eduardo Habkost, QEMU, Michael Walle, Gerd Hoffmann,
Paolo Bonzini
On Tue, Jul 9, 2019 at 10:58 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Finally add audiodev= options to audio frontends so users can specify
> which backend to use when multiple backends exist. Not specifying an
> audiodev= option currently causes the first audiodev to be used, this is
> fixed in the next commit.
>
> Example usage: -audiodev pa,id=foo -device AC97,audiodev=foo
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> audio/audio.h | 3 ++
> include/hw/qdev-properties.h | 3 ++
> hw/audio/ac97.c | 1 +
> hw/audio/adlib.c | 1 +
> hw/audio/cs4231a.c | 1 +
> hw/audio/es1370.c | 7 +++-
> hw/audio/gus.c | 1 +
> hw/audio/hda-codec.c | 1 +
> hw/audio/milkymist-ac97.c | 6 ++++
> hw/audio/pcspk.c | 1 +
> hw/audio/pl041.c | 1 +
> hw/audio/sb16.c | 1 +
> hw/audio/wm8750.c | 6 ++++
> hw/core/qdev-properties-system.c | 57 ++++++++++++++++++++++++++++++++
> hw/usb/dev-audio.c | 1 +
> 15 files changed, 90 insertions(+), 1 deletion(-)
>
> diff --git a/audio/audio.h b/audio/audio.h
> index ad2457f4de..c0722a5cda 100644
> --- a/audio/audio.h
> +++ b/audio/audio.h
> @@ -181,4 +181,7 @@ void audio_legacy_help(void);
> AudioState *audio_state_by_name(const char *name);
> const char *audio_get_id(QEMUSoundCard *card);
>
> +#define DEFINE_AUDIO_PROPERTIES(_s, _f) \
> + DEFINE_PROP_AUDIODEV("audiodev", _s, _f)
> +
> #endif /* QEMU_AUDIO_H */
> diff --git a/include/hw/qdev-properties.h b/include/hw/qdev-properties.h
> index 1eae5ab056..de0b2c8423 100644
> --- a/include/hw/qdev-properties.h
> +++ b/include/hw/qdev-properties.h
> @@ -35,6 +35,7 @@ extern const PropertyInfo qdev_prop_blocksize;
> extern const PropertyInfo qdev_prop_pci_host_devaddr;
> extern const PropertyInfo qdev_prop_uuid;
> extern const PropertyInfo qdev_prop_arraylen;
> +extern const PropertyInfo qdev_prop_audiodev;
> extern const PropertyInfo qdev_prop_link;
> extern const PropertyInfo qdev_prop_off_auto_pcibar;
> extern const PropertyInfo qdev_prop_pcie_link_speed;
> @@ -236,6 +237,8 @@ extern const PropertyInfo qdev_prop_pcie_link_width;
> + type_check(QemuUUID, typeof_field(_state, _field)), \
> .set_default = true, \
> }
> +#define DEFINE_PROP_AUDIODEV(_n, _s, _f) \
> + DEFINE_PROP(_n, _s, _f, qdev_prop_audiodev, QEMUSoundCard)
>
> #define DEFINE_PROP_END_OF_LIST() \
> {}
> diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
> index fb98da2678..0d8e524233 100644
> --- a/hw/audio/ac97.c
> +++ b/hw/audio/ac97.c
> @@ -1409,6 +1409,7 @@ static int ac97_init (PCIBus *bus)
> }
>
> static Property ac97_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(AC97LinkState, card),
> DEFINE_PROP_UINT32 ("use_broken_id", AC97LinkState, use_broken_id, 0),
> DEFINE_PROP_END_OF_LIST (),
> };
> diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
> index 7dd9a89b89..df2e781788 100644
> --- a/hw/audio/adlib.c
> +++ b/hw/audio/adlib.c
> @@ -299,6 +299,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
> }
>
> static Property adlib_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(AdlibState, card),
> DEFINE_PROP_UINT32 ("iobase", AdlibState, port, 0x220),
> DEFINE_PROP_UINT32 ("freq", AdlibState, freq, 44100),
> DEFINE_PROP_END_OF_LIST (),
> diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
> index 7216b41cc1..e3ea830b47 100644
> --- a/hw/audio/cs4231a.c
> +++ b/hw/audio/cs4231a.c
> @@ -689,6 +689,7 @@ static int cs4231a_init (ISABus *bus)
> }
>
> static Property cs4231a_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(CSState, card),
> DEFINE_PROP_UINT32 ("iobase", CSState, port, 0x534),
> DEFINE_PROP_UINT32 ("irq", CSState, irq, 9),
> DEFINE_PROP_UINT32 ("dma", CSState, dma, 3),
> diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
> index 260c142b70..7589671d20 100644
> --- a/hw/audio/es1370.c
> +++ b/hw/audio/es1370.c
> @@ -887,6 +887,11 @@ static int es1370_init (PCIBus *bus)
> return 0;
> }
>
> +static Property es1370_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(ES1370State, card),
> + DEFINE_PROP_END_OF_LIST(),
> +};
> +
> static void es1370_class_init (ObjectClass *klass, void *data)
> {
> DeviceClass *dc = DEVICE_CLASS (klass);
> @@ -903,6 +908,7 @@ static void es1370_class_init (ObjectClass *klass, void *data)
> dc->desc = "ENSONIQ AudioPCI ES1370";
> dc->vmsd = &vmstate_es1370;
> dc->reset = es1370_on_reset;
> + dc->props = es1370_properties;
> }
>
> static const TypeInfo es1370_info = {
> @@ -923,4 +929,3 @@ static void es1370_register_types (void)
> }
>
> type_init (es1370_register_types)
> -
> diff --git a/hw/audio/gus.c b/hw/audio/gus.c
> index 9ab51631d9..566864bc9e 100644
> --- a/hw/audio/gus.c
> +++ b/hw/audio/gus.c
> @@ -297,6 +297,7 @@ static int GUS_init (ISABus *bus)
> }
>
> static Property gus_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(GUSState, card),
> DEFINE_PROP_UINT32 ("freq", GUSState, freq, 44100),
> DEFINE_PROP_UINT32 ("iobase", GUSState, port, 0x240),
> DEFINE_PROP_UINT32 ("irq", GUSState, emu.gusirq, 7),
> diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
> index 84639b5424..967a10f189 100644
> --- a/hw/audio/hda-codec.c
> +++ b/hw/audio/hda-codec.c
> @@ -840,6 +840,7 @@ static const VMStateDescription vmstate_hda_audio = {
> };
>
> static Property hda_audio_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
> DEFINE_PROP_UINT32("debug", HDAAudioState, debug, 0),
> DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true),
> DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer, true),
> diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
> index bf6a5a6b96..4835229326 100644
> --- a/hw/audio/milkymist-ac97.c
> +++ b/hw/audio/milkymist-ac97.c
> @@ -329,6 +329,11 @@ static const VMStateDescription vmstate_milkymist_ac97 = {
> }
> };
>
> +static Property milkymist_ac97_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(MilkymistAC97State, card),
> + DEFINE_PROP_END_OF_LIST(),
> +};
> +
> static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
> {
> DeviceClass *dc = DEVICE_CLASS(klass);
> @@ -336,6 +341,7 @@ static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
> dc->realize = milkymist_ac97_realize;
> dc->reset = milkymist_ac97_reset;
> dc->vmsd = &vmstate_milkymist_ac97;
> + dc->props = milkymist_ac97_properties;
> }
>
> static const TypeInfo milkymist_ac97_info = {
> diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
> index a7b3ab0c2b..01127304c2 100644
> --- a/hw/audio/pcspk.c
> +++ b/hw/audio/pcspk.c
> @@ -209,6 +209,7 @@ static const VMStateDescription vmstate_spk = {
> };
>
> static Property pcspk_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(PCSpkState, card),
> DEFINE_PROP_UINT32("iobase", PCSpkState, iobase, -1),
> DEFINE_PROP_BOOL("migrate", PCSpkState, migrate, true),
> DEFINE_PROP_END_OF_LIST(),
> diff --git a/hw/audio/pl041.c b/hw/audio/pl041.c
> index 59c6ce1f2e..ce430a7821 100644
> --- a/hw/audio/pl041.c
> +++ b/hw/audio/pl041.c
> @@ -622,6 +622,7 @@ static const VMStateDescription vmstate_pl041 = {
> };
>
> static Property pl041_device_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(PL041State, codec.card),
> /* Non-compact FIFO depth property */
> DEFINE_PROP_UINT32("nc_fifo_depth", PL041State, fifo_depth,
> DEFAULT_FIFO_DEPTH),
> diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
> index a2db06a2bb..6b604979cf 100644
> --- a/hw/audio/sb16.c
> +++ b/hw/audio/sb16.c
> @@ -1421,6 +1421,7 @@ static int SB16_init (ISABus *bus)
> }
>
> static Property sb16_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(SB16State, card),
> DEFINE_PROP_UINT32 ("version", SB16State, ver, 0x0405), /* 4.5 */
> DEFINE_PROP_UINT32 ("iobase", SB16State, port, 0x220),
> DEFINE_PROP_UINT32 ("irq", SB16State, irq, 5),
> diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
> index 81f83eea88..dfb4156ff4 100644
> --- a/hw/audio/wm8750.c
> +++ b/hw/audio/wm8750.c
> @@ -701,6 +701,11 @@ void wm8750_set_bclk_in(void *opaque, int new_hz)
> wm8750_clk_update(s, 1);
> }
>
> +static Property wm8750_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(WM8750State, card),
> + DEFINE_PROP_END_OF_LIST(),
> +};
> +
> static void wm8750_class_init(ObjectClass *klass, void *data)
> {
> DeviceClass *dc = DEVICE_CLASS(klass);
> @@ -711,6 +716,7 @@ static void wm8750_class_init(ObjectClass *klass, void *data)
> sc->recv = wm8750_rx;
> sc->send = wm8750_tx;
> dc->vmsd = &vmstate_wm8750;
> + dc->props = wm8750_properties;
> }
>
> static const TypeInfo wm8750_info = {
> diff --git a/hw/core/qdev-properties-system.c b/hw/core/qdev-properties-system.c
> index ba412dd2ca..bf9974370c 100644
> --- a/hw/core/qdev-properties-system.c
> +++ b/hw/core/qdev-properties-system.c
> @@ -11,6 +11,7 @@
> */
>
> #include "qemu/osdep.h"
> +#include "audio/audio.h"
> #include "net/net.h"
> #include "hw/qdev.h"
> #include "qapi/error.h"
> @@ -353,6 +354,62 @@ const PropertyInfo qdev_prop_netdev = {
> };
>
>
> +/* --- audiodev --- */
> +static void get_audiodev(Object *obj, Visitor *v, const char* name,
> + void *opaque, Error **errp)
> +{
> + DeviceState *dev = DEVICE(obj);
> + Property *prop = opaque;
> + QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
> + char *p = g_strdup(audio_get_id(card));
> +
> + visit_type_str(v, name, &p, errp);
> + g_free(p);
> +}
> +
> +static void set_audiodev(Object *obj, Visitor *v, const char* name,
> + void *opaque, Error **errp)
> +{
> + DeviceState *dev = DEVICE(obj);
> + Property *prop = opaque;
> + QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
> + AudioState *state;
> + Error *local_err = NULL;
> + int err = 0;
> + char *str;
> +
> + if (dev->realized) {
> + qdev_prop_set_after_realize(dev, name, errp);
> + return;
> + }
> +
> + visit_type_str(v, name, &str, &local_err);
> + if (local_err) {
> + error_propagate(errp, local_err);
> + return;
> + }
> +
> + state = audio_state_by_name(str);
> +
> + if (!state) {
> + err = -ENOENT;
> + goto out;
> + }
> + card->state = state;
> +
> +out:
> + error_set_from_qdev_prop_error(errp, err, dev, prop, str);
> + g_free(str);
> +}
> +
> +const PropertyInfo qdev_prop_audiodev = {
> + .name = "str",
> + .description = "ID of an audiodev to use as a backend",
> + /* release done on shutdown */
> + .get = get_audiodev,
> + .set = set_audiodev,
> +};
> +
> void qdev_prop_set_drive(DeviceState *dev, const char *name,
> BlockBackend *value, Error **errp)
> {
> diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
> index 718ab53cbb..1b1cf91132 100644
> --- a/hw/usb/dev-audio.c
> +++ b/hw/usb/dev-audio.c
> @@ -666,6 +666,7 @@ static const VMStateDescription vmstate_usb_audio = {
> };
>
> static Property usb_audio_properties[] = {
> + DEFINE_AUDIO_PROPERTIES(USBAudioState, card),
> DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0),
> DEFINE_PROP_UINT32("buffer", USBAudioState, buffer,
> 32 * USBAUDIO_PACKET_SIZE),
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (2 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 03/11] audio: add audiodev properties to frontends Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:30 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 05/11] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
` (8 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This means you should probably stop using -soundhw (as it doesn't allow
you to specify any options) and add the device manually with -device.
The exception is pcspk, it's currently not possible to manually add it.
To use it with audiodev, use something like this:
-audiodev id=foo,... -global isa-pcspk.audiodev=foo -soundhw pcspk
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 24 ++--
audio/paaudio.c | 329 +++++++++++++++++++++++++++---------------------
2 files changed, 203 insertions(+), 150 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e9dd7c8b32..82dd0e3e13 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -101,6 +101,8 @@ const struct mixeng_volume nominal_volume = {
#endif
};
+static bool legacy_config = true;
+
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
@@ -1392,7 +1394,7 @@ static AudiodevListEntry *audiodev_find(
* if dev == NULL => legacy implicit initialization, return the already created
* state or create a new one
*/
-static AudioState *audio_init(Audiodev *dev)
+static AudioState *audio_init(Audiodev *dev, const char *name)
{
static bool atexit_registered;
size_t i;
@@ -1406,12 +1408,13 @@ static AudioState *audio_init(Audiodev *dev)
if (dev) {
/* -audiodev option */
+ legacy_config = false;
drvname = AudiodevDriver_str(dev->driver);
} else if (!QTAILQ_EMPTY(&audio_states)) {
- /*
- * todo: check for -audiodev once we have normal audiodev selection
- * support
- */
+ if (!legacy_config) {
+ dolog("You must specify an audiodev= for the device %s\n", name);
+ exit(1);
+ }
return QTAILQ_FIRST(&audio_states);
} else {
/* legacy implicit initialization */
@@ -1518,7 +1521,7 @@ void audio_free_audiodev_list(AudiodevListHead *head)
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
if (!card->state) {
- card->state = audio_init(NULL);
+ card->state = audio_init(NULL, name);
}
card->name = g_strdup (name);
@@ -1544,8 +1547,11 @@ CaptureVoiceOut *AUD_add_capture(
struct capture_callback *cb;
if (!s) {
- /* todo: remove when we have normal audiodev selection support */
- s = audio_init(NULL);
+ if (!legacy_config) {
+ dolog("You must specify audiodev when trying to capture\n");
+ return NULL;
+ }
+ s = audio_init(NULL, NULL);
}
if (audio_validate_settings (as)) {
@@ -1776,7 +1782,7 @@ void audio_init_audiodevs(void)
AudiodevListEntry *e;
QSIMPLEQ_FOREACH(e, &audiodevs, next) {
- audio_init(e->dev);
+ audio_init(e->dev, NULL);
}
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 5fc886bb33..cc3a34c2ea 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -11,10 +11,21 @@
#include "audio_int.h"
#include "audio_pt_int.h"
-typedef struct {
- Audiodev *dev;
+typedef struct PAConnection {
+ char *server;
+ int refcount;
+ QTAILQ_ENTRY(PAConnection) list;
+
pa_threaded_mainloop *mainloop;
pa_context *context;
+} PAConnection;
+
+static QTAILQ_HEAD(PAConnectionHead, PAConnection) pa_conns =
+ QTAILQ_HEAD_INITIALIZER(pa_conns);
+
+typedef struct {
+ Audiodev *dev;
+ PAConnection *conn;
} paaudio;
typedef struct {
@@ -45,7 +56,7 @@ typedef struct {
int samples;
} PAVoiceIn;
-static void qpa_audio_fini(void *opaque);
+static void qpa_conn_fini(PAConnection *c);
static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
{
@@ -108,11 +119,11 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
{
- paaudio *g = p->g;
+ PAConnection *c = p->g->conn;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
while (length > 0) {
size_t l;
@@ -121,11 +132,11 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
int r;
r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
if (!p->read_data) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ pa_threaded_mainloop_wait(c->mainloop);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
} else {
p->read_index = 0;
}
@@ -148,53 +159,53 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
p->read_length = 0;
p->read_index = 0;
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
}
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return 0;
unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return -1;
}
static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
{
- paaudio *g = p->g;
+ PAConnection *c = p->g->conn;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
while (length > 0) {
size_t l;
int r;
while (!(l = pa_stream_writable_size (p->stream))) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ pa_threaded_mainloop_wait(c->mainloop);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
}
- CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
if (l > length) {
l = length;
}
r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
data = (const uint8_t *) data + l;
length -= l;
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return 0;
unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return -1;
}
@@ -432,13 +443,13 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
static void context_state_cb (pa_context *c, void *userdata)
{
- paaudio *g = userdata;
+ PAConnection *conn = userdata;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(conn->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
@@ -451,14 +462,14 @@ static void context_state_cb (pa_context *c, void *userdata)
static void stream_state_cb (pa_stream *s, void * userdata)
{
- paaudio *g = userdata;
+ PAConnection *c = userdata;
switch (pa_stream_get_state (s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(c->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
@@ -469,13 +480,13 @@ static void stream_state_cb (pa_stream *s, void * userdata)
static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
{
- paaudio *g = userdata;
+ PAConnection *c = userdata;
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(c->mainloop, 0);
}
static pa_stream *qpa_simple_new (
- paaudio *g,
+ PAConnection *c,
const char *name,
pa_stream_direction_t dir,
const char *dev,
@@ -486,50 +497,48 @@ static pa_stream *qpa_simple_new (
{
int r;
pa_stream *stream;
+ pa_stream_flags_t flags;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new (g->context, name, ss, map);
+ stream = pa_stream_new(c->context, name, ss, map);
if (!stream) {
goto fail;
}
- pa_stream_set_state_callback (stream, stream_state_cb, g);
- pa_stream_set_read_callback (stream, stream_request_cb, g);
- pa_stream_set_write_callback (stream, stream_request_cb, g);
+ pa_stream_set_state_callback(stream, stream_state_cb, c);
+ pa_stream_set_read_callback(stream, stream_request_cb, c);
+ pa_stream_set_write_callback(stream, stream_request_cb, c);
+
+ flags =
+ PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+ | PA_STREAM_ADJUST_LATENCY
+#endif
+ | PA_STREAM_AUTO_TIMING_UPDATE;
if (dir == PA_STREAM_PLAYBACK) {
- r = pa_stream_connect_playback (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+ r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
} else {
- r = pa_stream_connect_record (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE);
+ r = pa_stream_connect_record(stream, dev, attr, flags);
}
if (r < 0) {
goto fail;
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return stream;
fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
if (stream) {
pa_stream_unref (stream);
}
- *rerror = pa_context_errno (g->context);
+ *rerror = pa_context_errno(c->context);
return NULL;
}
@@ -545,6 +554,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->out;
+ PAConnection *c = g->conn;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -558,7 +568,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- g,
+ c,
"qemu",
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
@@ -612,6 +622,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->in;
+ PAConnection *c = g->conn;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -625,7 +636,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- g,
+ c,
"qemu",
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
@@ -714,7 +725,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = pa->g;
+ PAConnection *c = pa->g->conn;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
@@ -734,28 +745,29 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- op = pa_context_set_sink_input_volume (g->context,
+ op = pa_context_set_sink_input_volume(c->context,
pa_stream_get_index (pa->stream),
&v, NULL, NULL);
- if (!op)
- qpa_logerr (pa_context_errno (g->context),
- "set_sink_input_volume() failed\n");
- else
- pa_operation_unref (op);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_volume() failed\n");
+ } else {
+ pa_operation_unref(op);
+ }
- op = pa_context_set_sink_input_mute (g->context,
+ op = pa_context_set_sink_input_mute(c->context,
pa_stream_get_index (pa->stream),
sw->vol.mute, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
- "set_sink_input_mute() failed\n");
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_mute() failed\n");
} else {
- pa_operation_unref (op);
+ pa_operation_unref(op);
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
}
}
return 0;
@@ -766,7 +778,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = pa->g;
+ PAConnection *c = pa->g->conn;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
@@ -786,29 +798,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- op = pa_context_set_source_output_volume (g->context,
- pa_stream_get_index (pa->stream),
+ op = pa_context_set_source_output_volume(c->context,
+ pa_stream_get_index(pa->stream),
&v, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
- "set_source_output_volume() failed\n");
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_volume() failed\n");
} else {
pa_operation_unref(op);
}
- op = pa_context_set_source_output_mute (g->context,
+ op = pa_context_set_source_output_mute(c->context,
pa_stream_get_index (pa->stream),
sw->vol.mute, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
- "set_source_output_mute() failed\n");
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_mute() failed\n");
} else {
pa_operation_unref (op);
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
}
}
return 0;
@@ -828,11 +840,75 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
return 1;
}
+/* common */
+static void *qpa_conn_init(const char *server)
+{
+ PAConnection *c = g_malloc0(sizeof(PAConnection));
+ QTAILQ_INSERT_TAIL(&pa_conns, c, list);
+
+ c->mainloop = pa_threaded_mainloop_new();
+ if (!c->mainloop) {
+ goto fail;
+ }
+
+ c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
+ server);
+ if (!c->context) {
+ goto fail;
+ }
+
+ pa_context_set_state_callback(c->context, context_state_cb, c);
+
+ if (pa_context_connect(c->context, server, 0, NULL) < 0) {
+ qpa_logerr(pa_context_errno(c->context),
+ "pa_context_connect() failed\n");
+ goto fail;
+ }
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ if (pa_threaded_mainloop_start(c->mainloop) < 0) {
+ goto unlock_and_fail;
+ }
+
+ for (;;) {
+ pa_context_state_t state;
+
+ state = pa_context_get_state(c->context);
+
+ if (state == PA_CONTEXT_READY) {
+ break;
+ }
+
+ if (!PA_CONTEXT_IS_GOOD(state)) {
+ qpa_logerr(pa_context_errno(c->context),
+ "Wrong context state\n");
+ goto unlock_and_fail;
+ }
+
+ /* Wait until the context is ready */
+ pa_threaded_mainloop_wait(c->mainloop);
+ }
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return c;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+fail:
+ AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+ qpa_conn_fini(c);
+ return NULL;
+}
+
static void *qpa_audio_init(Audiodev *dev)
{
paaudio *g;
AudiodevPaOptions *popts = &dev->u.pa;
const char *server;
+ PAConnection *c;
+
+ assert(dev->driver == AUDIODEV_DRIVER_PA);
if (!popts->has_server) {
char pidfile[64];
@@ -849,93 +925,64 @@ static void *qpa_audio_init(Audiodev *dev)
}
}
- assert(dev->driver == AUDIODEV_DRIVER_PA);
-
- g = g_malloc(sizeof(paaudio));
- server = popts->has_server ? popts->server : NULL;
-
if (!qpa_validate_per_direction_opts(dev, popts->in)) {
- goto fail;
+ return NULL;
}
if (!qpa_validate_per_direction_opts(dev, popts->out)) {
- goto fail;
+ return NULL;
}
+ g = g_malloc0(sizeof(paaudio));
+ server = popts->has_server ? popts->server : NULL;
+
g->dev = dev;
- g->mainloop = NULL;
- g->context = NULL;
- g->mainloop = pa_threaded_mainloop_new ();
- if (!g->mainloop) {
- goto fail;
- }
-
- g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
- server);
- if (!g->context) {
- goto fail;
- }
-
- pa_context_set_state_callback (g->context, context_state_cb, g);
-
- if (pa_context_connect(g->context, server, 0, NULL) < 0) {
- qpa_logerr (pa_context_errno (g->context),
- "pa_context_connect() failed\n");
- goto fail;
- }
-
- pa_threaded_mainloop_lock (g->mainloop);
-
- if (pa_threaded_mainloop_start (g->mainloop) < 0) {
- goto unlock_and_fail;
- }
-
- for (;;) {
- pa_context_state_t state;
-
- state = pa_context_get_state (g->context);
-
- if (state == PA_CONTEXT_READY) {
+ QTAILQ_FOREACH(c, &pa_conns, list) {
+ if (server == NULL || c->server == NULL ?
+ server == c->server :
+ strcmp(server, c->server) == 0) {
+ g->conn = c;
break;
}
-
- if (!PA_CONTEXT_IS_GOOD (state)) {
- qpa_logerr (pa_context_errno (g->context),
- "Wrong context state\n");
- goto unlock_and_fail;
- }
-
- /* Wait until the context is ready */
- pa_threaded_mainloop_wait (g->mainloop);
+ }
+ if (!g->conn) {
+ g->conn = qpa_conn_init(server);
+ }
+ if (!g->conn) {
+ g_free(g);
+ return NULL;
}
- pa_threaded_mainloop_unlock (g->mainloop);
-
+ ++g->conn->refcount;
return g;
+}
-unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
-fail:
- AUD_log (AUDIO_CAP, "Failed to initialize PA context");
- qpa_audio_fini(g);
- return NULL;
+static void qpa_conn_fini(PAConnection *c)
+{
+ if (c->mainloop) {
+ pa_threaded_mainloop_stop(c->mainloop);
+ }
+
+ if (c->context) {
+ pa_context_disconnect(c->context);
+ pa_context_unref(c->context);
+ }
+
+ if (c->mainloop) {
+ pa_threaded_mainloop_free(c->mainloop);
+ }
+
+ QTAILQ_REMOVE(&pa_conns, c, list);
+ g_free(c);
}
static void qpa_audio_fini (void *opaque)
{
paaudio *g = opaque;
+ PAConnection *c = g->conn;
- if (g->mainloop) {
- pa_threaded_mainloop_stop (g->mainloop);
- }
-
- if (g->context) {
- pa_context_disconnect (g->context);
- pa_context_unref (g->context);
- }
-
- if (g->mainloop) {
- pa_threaded_mainloop_free (g->mainloop);
+ if (--c->refcount == 0) {
+ qpa_conn_fini(c);
}
g_free(g);
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present
2019-07-09 18:35 ` [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
@ 2019-07-10 19:30 ` Marc-André Lureau
2019-07-11 19:09 ` Zoltán Kővágó
0 siblings, 1 reply; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:30 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 11:00 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> This means you should probably stop using -soundhw (as it doesn't allow
> you to specify any options) and add the device manually with -device.
> The exception is pcspk, it's currently not possible to manually add it.
> To use it with audiodev, use something like this:
>
> -audiodev id=foo,... -global isa-pcspk.audiodev=foo -soundhw pcspk
Can you seperate the paaudio changes to ease review?
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> audio/audio.c | 24 ++--
> audio/paaudio.c | 329 +++++++++++++++++++++++++++---------------------
> 2 files changed, 203 insertions(+), 150 deletions(-)
>
> diff --git a/audio/audio.c b/audio/audio.c
> index e9dd7c8b32..82dd0e3e13 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -101,6 +101,8 @@ const struct mixeng_volume nominal_volume = {
> #endif
> };
>
> +static bool legacy_config = true;
> +
> #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
> #error No its not
> #else
> @@ -1392,7 +1394,7 @@ static AudiodevListEntry *audiodev_find(
> * if dev == NULL => legacy implicit initialization, return the already created
> * state or create a new one
> */
> -static AudioState *audio_init(Audiodev *dev)
> +static AudioState *audio_init(Audiodev *dev, const char *name)
> {
> static bool atexit_registered;
> size_t i;
> @@ -1406,12 +1408,13 @@ static AudioState *audio_init(Audiodev *dev)
>
> if (dev) {
> /* -audiodev option */
> + legacy_config = false;
> drvname = AudiodevDriver_str(dev->driver);
> } else if (!QTAILQ_EMPTY(&audio_states)) {
> - /*
> - * todo: check for -audiodev once we have normal audiodev selection
> - * support
> - */
> + if (!legacy_config) {
> + dolog("You must specify an audiodev= for the device %s\n", name);
> + exit(1);
> + }
> return QTAILQ_FIRST(&audio_states);
> } else {
> /* legacy implicit initialization */
> @@ -1518,7 +1521,7 @@ void audio_free_audiodev_list(AudiodevListHead *head)
> void AUD_register_card (const char *name, QEMUSoundCard *card)
> {
> if (!card->state) {
> - card->state = audio_init(NULL);
> + card->state = audio_init(NULL, name);
> }
>
> card->name = g_strdup (name);
> @@ -1544,8 +1547,11 @@ CaptureVoiceOut *AUD_add_capture(
> struct capture_callback *cb;
>
> if (!s) {
> - /* todo: remove when we have normal audiodev selection support */
> - s = audio_init(NULL);
> + if (!legacy_config) {
> + dolog("You must specify audiodev when trying to capture\n");
> + return NULL;
> + }
> + s = audio_init(NULL, NULL);
> }
>
> if (audio_validate_settings (as)) {
> @@ -1776,7 +1782,7 @@ void audio_init_audiodevs(void)
> AudiodevListEntry *e;
>
> QSIMPLEQ_FOREACH(e, &audiodevs, next) {
> - audio_init(e->dev);
> + audio_init(e->dev, NULL);
> }
> }
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index 5fc886bb33..cc3a34c2ea 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -11,10 +11,21 @@
> #include "audio_int.h"
> #include "audio_pt_int.h"
>
> -typedef struct {
> - Audiodev *dev;
> +typedef struct PAConnection {
> + char *server;
> + int refcount;
> + QTAILQ_ENTRY(PAConnection) list;
> +
> pa_threaded_mainloop *mainloop;
> pa_context *context;
> +} PAConnection;
> +
> +static QTAILQ_HEAD(PAConnectionHead, PAConnection) pa_conns =
> + QTAILQ_HEAD_INITIALIZER(pa_conns);
> +
> +typedef struct {
> + Audiodev *dev;
> + PAConnection *conn;
> } paaudio;
>
> typedef struct {
> @@ -45,7 +56,7 @@ typedef struct {
> int samples;
> } PAVoiceIn;
>
> -static void qpa_audio_fini(void *opaque);
> +static void qpa_conn_fini(PAConnection *c);
>
> static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
> {
> @@ -108,11 +119,11 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
>
> static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
> {
> - paaudio *g = p->g;
> + PAConnection *c = p->g->conn;
>
> - pa_threaded_mainloop_lock (g->mainloop);
> + pa_threaded_mainloop_lock(c->mainloop);
>
> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>
> while (length > 0) {
> size_t l;
> @@ -121,11 +132,11 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
> int r;
>
> r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
> - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
> + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
>
> if (!p->read_data) {
> - pa_threaded_mainloop_wait (g->mainloop);
> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
> + pa_threaded_mainloop_wait(c->mainloop);
> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
> } else {
> p->read_index = 0;
> }
> @@ -148,53 +159,53 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
> p->read_length = 0;
> p->read_index = 0;
>
> - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
> + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
> }
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> return 0;
>
> unlock_and_fail:
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> return -1;
> }
>
> static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
> {
> - paaudio *g = p->g;
> + PAConnection *c = p->g->conn;
>
> - pa_threaded_mainloop_lock (g->mainloop);
> + pa_threaded_mainloop_lock(c->mainloop);
>
> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>
> while (length > 0) {
> size_t l;
> int r;
>
> while (!(l = pa_stream_writable_size (p->stream))) {
> - pa_threaded_mainloop_wait (g->mainloop);
> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
> + pa_threaded_mainloop_wait(c->mainloop);
> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
> }
>
> - CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
> + CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
>
> if (l > length) {
> l = length;
> }
>
> r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
> - CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
> + CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
>
> data = (const uint8_t *) data + l;
> length -= l;
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> return 0;
>
> unlock_and_fail:
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> return -1;
> }
>
> @@ -432,13 +443,13 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
>
> static void context_state_cb (pa_context *c, void *userdata)
> {
> - paaudio *g = userdata;
> + PAConnection *conn = userdata;
>
> switch (pa_context_get_state(c)) {
> case PA_CONTEXT_READY:
> case PA_CONTEXT_TERMINATED:
> case PA_CONTEXT_FAILED:
> - pa_threaded_mainloop_signal (g->mainloop, 0);
> + pa_threaded_mainloop_signal(conn->mainloop, 0);
> break;
>
> case PA_CONTEXT_UNCONNECTED:
> @@ -451,14 +462,14 @@ static void context_state_cb (pa_context *c, void *userdata)
>
> static void stream_state_cb (pa_stream *s, void * userdata)
> {
> - paaudio *g = userdata;
> + PAConnection *c = userdata;
>
> switch (pa_stream_get_state (s)) {
>
> case PA_STREAM_READY:
> case PA_STREAM_FAILED:
> case PA_STREAM_TERMINATED:
> - pa_threaded_mainloop_signal (g->mainloop, 0);
> + pa_threaded_mainloop_signal(c->mainloop, 0);
> break;
>
> case PA_STREAM_UNCONNECTED:
> @@ -469,13 +480,13 @@ static void stream_state_cb (pa_stream *s, void * userdata)
>
> static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
> {
> - paaudio *g = userdata;
> + PAConnection *c = userdata;
>
> - pa_threaded_mainloop_signal (g->mainloop, 0);
> + pa_threaded_mainloop_signal(c->mainloop, 0);
> }
>
> static pa_stream *qpa_simple_new (
> - paaudio *g,
> + PAConnection *c,
> const char *name,
> pa_stream_direction_t dir,
> const char *dev,
> @@ -486,50 +497,48 @@ static pa_stream *qpa_simple_new (
> {
> int r;
> pa_stream *stream;
> + pa_stream_flags_t flags;
>
> - pa_threaded_mainloop_lock (g->mainloop);
> + pa_threaded_mainloop_lock(c->mainloop);
>
> - stream = pa_stream_new (g->context, name, ss, map);
> + stream = pa_stream_new(c->context, name, ss, map);
> if (!stream) {
> goto fail;
> }
>
> - pa_stream_set_state_callback (stream, stream_state_cb, g);
> - pa_stream_set_read_callback (stream, stream_request_cb, g);
> - pa_stream_set_write_callback (stream, stream_request_cb, g);
> + pa_stream_set_state_callback(stream, stream_state_cb, c);
> + pa_stream_set_read_callback(stream, stream_request_cb, c);
> + pa_stream_set_write_callback(stream, stream_request_cb, c);
> +
> + flags =
> + PA_STREAM_INTERPOLATE_TIMING
> +#ifdef PA_STREAM_ADJUST_LATENCY
> + | PA_STREAM_ADJUST_LATENCY
> +#endif
> + | PA_STREAM_AUTO_TIMING_UPDATE;
>
> if (dir == PA_STREAM_PLAYBACK) {
> - r = pa_stream_connect_playback (stream, dev, attr,
> - PA_STREAM_INTERPOLATE_TIMING
> -#ifdef PA_STREAM_ADJUST_LATENCY
> - |PA_STREAM_ADJUST_LATENCY
> -#endif
> - |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
> + r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
> } else {
> - r = pa_stream_connect_record (stream, dev, attr,
> - PA_STREAM_INTERPOLATE_TIMING
> -#ifdef PA_STREAM_ADJUST_LATENCY
> - |PA_STREAM_ADJUST_LATENCY
> -#endif
> - |PA_STREAM_AUTO_TIMING_UPDATE);
> + r = pa_stream_connect_record(stream, dev, attr, flags);
> }
>
> if (r < 0) {
> goto fail;
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
>
> return stream;
>
> fail:
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
>
> if (stream) {
> pa_stream_unref (stream);
> }
>
> - *rerror = pa_context_errno (g->context);
> + *rerror = pa_context_errno(c->context);
>
> return NULL;
> }
> @@ -545,6 +554,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
> paaudio *g = pa->g = drv_opaque;
> AudiodevPaOptions *popts = &g->dev->u.pa;
> AudiodevPaPerDirectionOptions *ppdo = popts->out;
> + PAConnection *c = g->conn;
>
> ss.format = audfmt_to_pa (as->fmt, as->endianness);
> ss.channels = as->nchannels;
> @@ -558,7 +568,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
> obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
>
> pa->stream = qpa_simple_new (
> - g,
> + c,
> "qemu",
> PA_STREAM_PLAYBACK,
> ppdo->has_name ? ppdo->name : NULL,
> @@ -612,6 +622,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> paaudio *g = pa->g = drv_opaque;
> AudiodevPaOptions *popts = &g->dev->u.pa;
> AudiodevPaPerDirectionOptions *ppdo = popts->in;
> + PAConnection *c = g->conn;
>
> ss.format = audfmt_to_pa (as->fmt, as->endianness);
> ss.channels = as->nchannels;
> @@ -625,7 +636,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
>
> pa->stream = qpa_simple_new (
> - g,
> + c,
> "qemu",
> PA_STREAM_RECORD,
> ppdo->has_name ? ppdo->name : NULL,
> @@ -714,7 +725,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
> PAVoiceOut *pa = (PAVoiceOut *) hw;
> pa_operation *op;
> pa_cvolume v;
> - paaudio *g = pa->g;
> + PAConnection *c = pa->g->conn;
>
> #ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
> pa_cvolume_init (&v); /* function is present in 0.9.13+ */
> @@ -734,28 +745,29 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
> v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
> v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
>
> - pa_threaded_mainloop_lock (g->mainloop);
> + pa_threaded_mainloop_lock(c->mainloop);
>
> - op = pa_context_set_sink_input_volume (g->context,
> + op = pa_context_set_sink_input_volume(c->context,
> pa_stream_get_index (pa->stream),
> &v, NULL, NULL);
> - if (!op)
> - qpa_logerr (pa_context_errno (g->context),
> - "set_sink_input_volume() failed\n");
> - else
> - pa_operation_unref (op);
> + if (!op) {
> + qpa_logerr(pa_context_errno(c->context),
> + "set_sink_input_volume() failed\n");
> + } else {
> + pa_operation_unref(op);
> + }
>
> - op = pa_context_set_sink_input_mute (g->context,
> + op = pa_context_set_sink_input_mute(c->context,
> pa_stream_get_index (pa->stream),
> sw->vol.mute, NULL, NULL);
> if (!op) {
> - qpa_logerr (pa_context_errno (g->context),
> - "set_sink_input_mute() failed\n");
> + qpa_logerr(pa_context_errno(c->context),
> + "set_sink_input_mute() failed\n");
> } else {
> - pa_operation_unref (op);
> + pa_operation_unref(op);
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> }
> }
> return 0;
> @@ -766,7 +778,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
> PAVoiceIn *pa = (PAVoiceIn *) hw;
> pa_operation *op;
> pa_cvolume v;
> - paaudio *g = pa->g;
> + PAConnection *c = pa->g->conn;
>
> #ifdef PA_CHECK_VERSION
> pa_cvolume_init (&v);
> @@ -786,29 +798,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
> v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
> v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
>
> - pa_threaded_mainloop_lock (g->mainloop);
> + pa_threaded_mainloop_lock(c->mainloop);
>
> - op = pa_context_set_source_output_volume (g->context,
> - pa_stream_get_index (pa->stream),
> + op = pa_context_set_source_output_volume(c->context,
> + pa_stream_get_index(pa->stream),
> &v, NULL, NULL);
> if (!op) {
> - qpa_logerr (pa_context_errno (g->context),
> - "set_source_output_volume() failed\n");
> + qpa_logerr(pa_context_errno(c->context),
> + "set_source_output_volume() failed\n");
> } else {
> pa_operation_unref(op);
> }
>
> - op = pa_context_set_source_output_mute (g->context,
> + op = pa_context_set_source_output_mute(c->context,
> pa_stream_get_index (pa->stream),
> sw->vol.mute, NULL, NULL);
> if (!op) {
> - qpa_logerr (pa_context_errno (g->context),
> - "set_source_output_mute() failed\n");
> + qpa_logerr(pa_context_errno(c->context),
> + "set_source_output_mute() failed\n");
> } else {
> pa_operation_unref (op);
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> + pa_threaded_mainloop_unlock(c->mainloop);
> }
> }
> return 0;
> @@ -828,11 +840,75 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
> return 1;
> }
>
> +/* common */
> +static void *qpa_conn_init(const char *server)
> +{
> + PAConnection *c = g_malloc0(sizeof(PAConnection));
> + QTAILQ_INSERT_TAIL(&pa_conns, c, list);
> +
> + c->mainloop = pa_threaded_mainloop_new();
> + if (!c->mainloop) {
> + goto fail;
> + }
> +
> + c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
> + server);
> + if (!c->context) {
> + goto fail;
> + }
> +
> + pa_context_set_state_callback(c->context, context_state_cb, c);
> +
> + if (pa_context_connect(c->context, server, 0, NULL) < 0) {
> + qpa_logerr(pa_context_errno(c->context),
> + "pa_context_connect() failed\n");
> + goto fail;
> + }
> +
> + pa_threaded_mainloop_lock(c->mainloop);
> +
> + if (pa_threaded_mainloop_start(c->mainloop) < 0) {
> + goto unlock_and_fail;
> + }
> +
> + for (;;) {
> + pa_context_state_t state;
> +
> + state = pa_context_get_state(c->context);
> +
> + if (state == PA_CONTEXT_READY) {
> + break;
> + }
> +
> + if (!PA_CONTEXT_IS_GOOD(state)) {
> + qpa_logerr(pa_context_errno(c->context),
> + "Wrong context state\n");
> + goto unlock_and_fail;
> + }
> +
> + /* Wait until the context is ready */
> + pa_threaded_mainloop_wait(c->mainloop);
> + }
> +
> + pa_threaded_mainloop_unlock(c->mainloop);
> + return c;
> +
> +unlock_and_fail:
> + pa_threaded_mainloop_unlock(c->mainloop);
> +fail:
> + AUD_log (AUDIO_CAP, "Failed to initialize PA context");
> + qpa_conn_fini(c);
> + return NULL;
> +}
> +
> static void *qpa_audio_init(Audiodev *dev)
> {
> paaudio *g;
> AudiodevPaOptions *popts = &dev->u.pa;
> const char *server;
> + PAConnection *c;
> +
> + assert(dev->driver == AUDIODEV_DRIVER_PA);
>
> if (!popts->has_server) {
> char pidfile[64];
> @@ -849,93 +925,64 @@ static void *qpa_audio_init(Audiodev *dev)
> }
> }
>
> - assert(dev->driver == AUDIODEV_DRIVER_PA);
> -
> - g = g_malloc(sizeof(paaudio));
> - server = popts->has_server ? popts->server : NULL;
> -
> if (!qpa_validate_per_direction_opts(dev, popts->in)) {
> - goto fail;
> + return NULL;
> }
> if (!qpa_validate_per_direction_opts(dev, popts->out)) {
> - goto fail;
> + return NULL;
> }
>
> + g = g_malloc0(sizeof(paaudio));
> + server = popts->has_server ? popts->server : NULL;
> +
> g->dev = dev;
> - g->mainloop = NULL;
> - g->context = NULL;
>
> - g->mainloop = pa_threaded_mainloop_new ();
> - if (!g->mainloop) {
> - goto fail;
> - }
> -
> - g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
> - server);
> - if (!g->context) {
> - goto fail;
> - }
> -
> - pa_context_set_state_callback (g->context, context_state_cb, g);
> -
> - if (pa_context_connect(g->context, server, 0, NULL) < 0) {
> - qpa_logerr (pa_context_errno (g->context),
> - "pa_context_connect() failed\n");
> - goto fail;
> - }
> -
> - pa_threaded_mainloop_lock (g->mainloop);
> -
> - if (pa_threaded_mainloop_start (g->mainloop) < 0) {
> - goto unlock_and_fail;
> - }
> -
> - for (;;) {
> - pa_context_state_t state;
> -
> - state = pa_context_get_state (g->context);
> -
> - if (state == PA_CONTEXT_READY) {
> + QTAILQ_FOREACH(c, &pa_conns, list) {
> + if (server == NULL || c->server == NULL ?
> + server == c->server :
> + strcmp(server, c->server) == 0) {
> + g->conn = c;
> break;
> }
> -
> - if (!PA_CONTEXT_IS_GOOD (state)) {
> - qpa_logerr (pa_context_errno (g->context),
> - "Wrong context state\n");
> - goto unlock_and_fail;
> - }
> -
> - /* Wait until the context is ready */
> - pa_threaded_mainloop_wait (g->mainloop);
> + }
> + if (!g->conn) {
> + g->conn = qpa_conn_init(server);
> + }
> + if (!g->conn) {
> + g_free(g);
> + return NULL;
> }
>
> - pa_threaded_mainloop_unlock (g->mainloop);
> -
> + ++g->conn->refcount;
> return g;
> +}
>
> -unlock_and_fail:
> - pa_threaded_mainloop_unlock (g->mainloop);
> -fail:
> - AUD_log (AUDIO_CAP, "Failed to initialize PA context");
> - qpa_audio_fini(g);
> - return NULL;
> +static void qpa_conn_fini(PAConnection *c)
> +{
> + if (c->mainloop) {
> + pa_threaded_mainloop_stop(c->mainloop);
> + }
> +
> + if (c->context) {
> + pa_context_disconnect(c->context);
> + pa_context_unref(c->context);
> + }
> +
> + if (c->mainloop) {
> + pa_threaded_mainloop_free(c->mainloop);
> + }
> +
> + QTAILQ_REMOVE(&pa_conns, c, list);
> + g_free(c);
> }
>
> static void qpa_audio_fini (void *opaque)
> {
> paaudio *g = opaque;
> + PAConnection *c = g->conn;
>
> - if (g->mainloop) {
> - pa_threaded_mainloop_stop (g->mainloop);
> - }
> -
> - if (g->context) {
> - pa_context_disconnect (g->context);
> - pa_context_unref (g->context);
> - }
> -
> - if (g->mainloop) {
> - pa_threaded_mainloop_free (g->mainloop);
> + if (--c->refcount == 0) {
> + qpa_conn_fini(c);
> }
>
> g_free(g);
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present
2019-07-10 19:30 ` Marc-André Lureau
@ 2019-07-11 19:09 ` Zoltán Kővágó
0 siblings, 0 replies; 32+ messages in thread
From: Zoltán Kővágó @ 2019-07-11 19:09 UTC (permalink / raw)
To: Marc-André Lureau; +Cc: QEMU, Gerd Hoffmann
On 2019-07-10 21:30, Marc-André Lureau wrote:
> On Tue, Jul 9, 2019 at 11:00 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>>
>> This means you should probably stop using -soundhw (as it doesn't allow
>> you to specify any options) and add the device manually with -device.
>> The exception is pcspk, it's currently not possible to manually add it.
>> To use it with audiodev, use something like this:
>>
>> -audiodev id=foo,... -global isa-pcspk.audiodev=foo -soundhw pcspk
>
> Can you seperate the paaudio changes to ease review?
I'll look into it, unfortunately the two changes are more intermixed
than I'd like them to be.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> audio/audio.c | 24 ++--
>> audio/paaudio.c | 329 +++++++++++++++++++++++++++---------------------
>> 2 files changed, 203 insertions(+), 150 deletions(-)
>>
>> diff --git a/audio/audio.c b/audio/audio.c
>> index e9dd7c8b32..82dd0e3e13 100644
>> --- a/audio/audio.c
>> +++ b/audio/audio.c
>> @@ -101,6 +101,8 @@ const struct mixeng_volume nominal_volume = {
>> #endif
>> };
>>
>> +static bool legacy_config = true;
>> +
>> #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
>> #error No its not
>> #else
>> @@ -1392,7 +1394,7 @@ static AudiodevListEntry *audiodev_find(
>> * if dev == NULL => legacy implicit initialization, return the already created
>> * state or create a new one
>> */
>> -static AudioState *audio_init(Audiodev *dev)
>> +static AudioState *audio_init(Audiodev *dev, const char *name)
>> {
>> static bool atexit_registered;
>> size_t i;
>> @@ -1406,12 +1408,13 @@ static AudioState *audio_init(Audiodev *dev)
>>
>> if (dev) {
>> /* -audiodev option */
>> + legacy_config = false;
>> drvname = AudiodevDriver_str(dev->driver);
>> } else if (!QTAILQ_EMPTY(&audio_states)) {
>> - /*
>> - * todo: check for -audiodev once we have normal audiodev selection
>> - * support
>> - */
>> + if (!legacy_config) {
>> + dolog("You must specify an audiodev= for the device %s\n", name);
>> + exit(1);
>> + }
>> return QTAILQ_FIRST(&audio_states);
>> } else {
>> /* legacy implicit initialization */
>> @@ -1518,7 +1521,7 @@ void audio_free_audiodev_list(AudiodevListHead *head)
>> void AUD_register_card (const char *name, QEMUSoundCard *card)
>> {
>> if (!card->state) {
>> - card->state = audio_init(NULL);
>> + card->state = audio_init(NULL, name);
>> }
>>
>> card->name = g_strdup (name);
>> @@ -1544,8 +1547,11 @@ CaptureVoiceOut *AUD_add_capture(
>> struct capture_callback *cb;
>>
>> if (!s) {
>> - /* todo: remove when we have normal audiodev selection support */
>> - s = audio_init(NULL);
>> + if (!legacy_config) {
>> + dolog("You must specify audiodev when trying to capture\n");
>> + return NULL;
>> + }
>> + s = audio_init(NULL, NULL);
>> }
>>
>> if (audio_validate_settings (as)) {
>> @@ -1776,7 +1782,7 @@ void audio_init_audiodevs(void)
>> AudiodevListEntry *e;
>>
>> QSIMPLEQ_FOREACH(e, &audiodevs, next) {
>> - audio_init(e->dev);
>> + audio_init(e->dev, NULL);
>> }
>> }
>>
>> diff --git a/audio/paaudio.c b/audio/paaudio.c
>> index 5fc886bb33..cc3a34c2ea 100644
>> --- a/audio/paaudio.c
>> +++ b/audio/paaudio.c
>> @@ -11,10 +11,21 @@
>> #include "audio_int.h"
>> #include "audio_pt_int.h"
>>
>> -typedef struct {
>> - Audiodev *dev;
>> +typedef struct PAConnection {
>> + char *server;
>> + int refcount;
>> + QTAILQ_ENTRY(PAConnection) list;
>> +
>> pa_threaded_mainloop *mainloop;
>> pa_context *context;
>> +} PAConnection;
>> +
>> +static QTAILQ_HEAD(PAConnectionHead, PAConnection) pa_conns =
>> + QTAILQ_HEAD_INITIALIZER(pa_conns);
>> +
>> +typedef struct {
>> + Audiodev *dev;
>> + PAConnection *conn;
>> } paaudio;
>>
>> typedef struct {
>> @@ -45,7 +56,7 @@ typedef struct {
>> int samples;
>> } PAVoiceIn;
>>
>> -static void qpa_audio_fini(void *opaque);
>> +static void qpa_conn_fini(PAConnection *c);
>>
>> static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
>> {
>> @@ -108,11 +119,11 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
>>
>> static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
>> {
>> - paaudio *g = p->g;
>> + PAConnection *c = p->g->conn;
>>
>> - pa_threaded_mainloop_lock (g->mainloop);
>> + pa_threaded_mainloop_lock(c->mainloop);
>>
>> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
>> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>>
>> while (length > 0) {
>> size_t l;
>> @@ -121,11 +132,11 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
>> int r;
>>
>> r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
>> - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
>> + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
>>
>> if (!p->read_data) {
>> - pa_threaded_mainloop_wait (g->mainloop);
>> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
>> + pa_threaded_mainloop_wait(c->mainloop);
>> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>> } else {
>> p->read_index = 0;
>> }
>> @@ -148,53 +159,53 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
>> p->read_length = 0;
>> p->read_index = 0;
>>
>> - CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
>> + CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
>> }
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> return 0;
>>
>> unlock_and_fail:
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> return -1;
>> }
>>
>> static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
>> {
>> - paaudio *g = p->g;
>> + PAConnection *c = p->g->conn;
>>
>> - pa_threaded_mainloop_lock (g->mainloop);
>> + pa_threaded_mainloop_lock(c->mainloop);
>>
>> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
>> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>>
>> while (length > 0) {
>> size_t l;
>> int r;
>>
>> while (!(l = pa_stream_writable_size (p->stream))) {
>> - pa_threaded_mainloop_wait (g->mainloop);
>> - CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
>> + pa_threaded_mainloop_wait(c->mainloop);
>> + CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
>> }
>>
>> - CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
>> + CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
>>
>> if (l > length) {
>> l = length;
>> }
>>
>> r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
>> - CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
>> + CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
>>
>> data = (const uint8_t *) data + l;
>> length -= l;
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> return 0;
>>
>> unlock_and_fail:
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> return -1;
>> }
>>
>> @@ -432,13 +443,13 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
>>
>> static void context_state_cb (pa_context *c, void *userdata)
>> {
>> - paaudio *g = userdata;
>> + PAConnection *conn = userdata;
>>
>> switch (pa_context_get_state(c)) {
>> case PA_CONTEXT_READY:
>> case PA_CONTEXT_TERMINATED:
>> case PA_CONTEXT_FAILED:
>> - pa_threaded_mainloop_signal (g->mainloop, 0);
>> + pa_threaded_mainloop_signal(conn->mainloop, 0);
>> break;
>>
>> case PA_CONTEXT_UNCONNECTED:
>> @@ -451,14 +462,14 @@ static void context_state_cb (pa_context *c, void *userdata)
>>
>> static void stream_state_cb (pa_stream *s, void * userdata)
>> {
>> - paaudio *g = userdata;
>> + PAConnection *c = userdata;
>>
>> switch (pa_stream_get_state (s)) {
>>
>> case PA_STREAM_READY:
>> case PA_STREAM_FAILED:
>> case PA_STREAM_TERMINATED:
>> - pa_threaded_mainloop_signal (g->mainloop, 0);
>> + pa_threaded_mainloop_signal(c->mainloop, 0);
>> break;
>>
>> case PA_STREAM_UNCONNECTED:
>> @@ -469,13 +480,13 @@ static void stream_state_cb (pa_stream *s, void * userdata)
>>
>> static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
>> {
>> - paaudio *g = userdata;
>> + PAConnection *c = userdata;
>>
>> - pa_threaded_mainloop_signal (g->mainloop, 0);
>> + pa_threaded_mainloop_signal(c->mainloop, 0);
>> }
>>
>> static pa_stream *qpa_simple_new (
>> - paaudio *g,
>> + PAConnection *c,
>> const char *name,
>> pa_stream_direction_t dir,
>> const char *dev,
>> @@ -486,50 +497,48 @@ static pa_stream *qpa_simple_new (
>> {
>> int r;
>> pa_stream *stream;
>> + pa_stream_flags_t flags;
>>
>> - pa_threaded_mainloop_lock (g->mainloop);
>> + pa_threaded_mainloop_lock(c->mainloop);
>>
>> - stream = pa_stream_new (g->context, name, ss, map);
>> + stream = pa_stream_new(c->context, name, ss, map);
>> if (!stream) {
>> goto fail;
>> }
>>
>> - pa_stream_set_state_callback (stream, stream_state_cb, g);
>> - pa_stream_set_read_callback (stream, stream_request_cb, g);
>> - pa_stream_set_write_callback (stream, stream_request_cb, g);
>> + pa_stream_set_state_callback(stream, stream_state_cb, c);
>> + pa_stream_set_read_callback(stream, stream_request_cb, c);
>> + pa_stream_set_write_callback(stream, stream_request_cb, c);
>> +
>> + flags =
>> + PA_STREAM_INTERPOLATE_TIMING
>> +#ifdef PA_STREAM_ADJUST_LATENCY
>> + | PA_STREAM_ADJUST_LATENCY
>> +#endif
>> + | PA_STREAM_AUTO_TIMING_UPDATE;
>>
>> if (dir == PA_STREAM_PLAYBACK) {
>> - r = pa_stream_connect_playback (stream, dev, attr,
>> - PA_STREAM_INTERPOLATE_TIMING
>> -#ifdef PA_STREAM_ADJUST_LATENCY
>> - |PA_STREAM_ADJUST_LATENCY
>> -#endif
>> - |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
>> + r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
>> } else {
>> - r = pa_stream_connect_record (stream, dev, attr,
>> - PA_STREAM_INTERPOLATE_TIMING
>> -#ifdef PA_STREAM_ADJUST_LATENCY
>> - |PA_STREAM_ADJUST_LATENCY
>> -#endif
>> - |PA_STREAM_AUTO_TIMING_UPDATE);
>> + r = pa_stream_connect_record(stream, dev, attr, flags);
>> }
>>
>> if (r < 0) {
>> goto fail;
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>>
>> return stream;
>>
>> fail:
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>>
>> if (stream) {
>> pa_stream_unref (stream);
>> }
>>
>> - *rerror = pa_context_errno (g->context);
>> + *rerror = pa_context_errno(c->context);
>>
>> return NULL;
>> }
>> @@ -545,6 +554,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
>> paaudio *g = pa->g = drv_opaque;
>> AudiodevPaOptions *popts = &g->dev->u.pa;
>> AudiodevPaPerDirectionOptions *ppdo = popts->out;
>> + PAConnection *c = g->conn;
>>
>> ss.format = audfmt_to_pa (as->fmt, as->endianness);
>> ss.channels = as->nchannels;
>> @@ -558,7 +568,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
>> obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
>>
>> pa->stream = qpa_simple_new (
>> - g,
>> + c,
>> "qemu",
>> PA_STREAM_PLAYBACK,
>> ppdo->has_name ? ppdo->name : NULL,
>> @@ -612,6 +622,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>> paaudio *g = pa->g = drv_opaque;
>> AudiodevPaOptions *popts = &g->dev->u.pa;
>> AudiodevPaPerDirectionOptions *ppdo = popts->in;
>> + PAConnection *c = g->conn;
>>
>> ss.format = audfmt_to_pa (as->fmt, as->endianness);
>> ss.channels = as->nchannels;
>> @@ -625,7 +636,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>> obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
>>
>> pa->stream = qpa_simple_new (
>> - g,
>> + c,
>> "qemu",
>> PA_STREAM_RECORD,
>> ppdo->has_name ? ppdo->name : NULL,
>> @@ -714,7 +725,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
>> PAVoiceOut *pa = (PAVoiceOut *) hw;
>> pa_operation *op;
>> pa_cvolume v;
>> - paaudio *g = pa->g;
>> + PAConnection *c = pa->g->conn;
>>
>> #ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
>> pa_cvolume_init (&v); /* function is present in 0.9.13+ */
>> @@ -734,28 +745,29 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
>> v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
>> v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
>>
>> - pa_threaded_mainloop_lock (g->mainloop);
>> + pa_threaded_mainloop_lock(c->mainloop);
>>
>> - op = pa_context_set_sink_input_volume (g->context,
>> + op = pa_context_set_sink_input_volume(c->context,
>> pa_stream_get_index (pa->stream),
>> &v, NULL, NULL);
>> - if (!op)
>> - qpa_logerr (pa_context_errno (g->context),
>> - "set_sink_input_volume() failed\n");
>> - else
>> - pa_operation_unref (op);
>> + if (!op) {
>> + qpa_logerr(pa_context_errno(c->context),
>> + "set_sink_input_volume() failed\n");
>> + } else {
>> + pa_operation_unref(op);
>> + }
>>
>> - op = pa_context_set_sink_input_mute (g->context,
>> + op = pa_context_set_sink_input_mute(c->context,
>> pa_stream_get_index (pa->stream),
>> sw->vol.mute, NULL, NULL);
>> if (!op) {
>> - qpa_logerr (pa_context_errno (g->context),
>> - "set_sink_input_mute() failed\n");
>> + qpa_logerr(pa_context_errno(c->context),
>> + "set_sink_input_mute() failed\n");
>> } else {
>> - pa_operation_unref (op);
>> + pa_operation_unref(op);
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> }
>> }
>> return 0;
>> @@ -766,7 +778,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> PAVoiceIn *pa = (PAVoiceIn *) hw;
>> pa_operation *op;
>> pa_cvolume v;
>> - paaudio *g = pa->g;
>> + PAConnection *c = pa->g->conn;
>>
>> #ifdef PA_CHECK_VERSION
>> pa_cvolume_init (&v);
>> @@ -786,29 +798,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
>> v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
>>
>> - pa_threaded_mainloop_lock (g->mainloop);
>> + pa_threaded_mainloop_lock(c->mainloop);
>>
>> - op = pa_context_set_source_output_volume (g->context,
>> - pa_stream_get_index (pa->stream),
>> + op = pa_context_set_source_output_volume(c->context,
>> + pa_stream_get_index(pa->stream),
>> &v, NULL, NULL);
>> if (!op) {
>> - qpa_logerr (pa_context_errno (g->context),
>> - "set_source_output_volume() failed\n");
>> + qpa_logerr(pa_context_errno(c->context),
>> + "set_source_output_volume() failed\n");
>> } else {
>> pa_operation_unref(op);
>> }
>>
>> - op = pa_context_set_source_output_mute (g->context,
>> + op = pa_context_set_source_output_mute(c->context,
>> pa_stream_get_index (pa->stream),
>> sw->vol.mute, NULL, NULL);
>> if (!op) {
>> - qpa_logerr (pa_context_errno (g->context),
>> - "set_source_output_mute() failed\n");
>> + qpa_logerr(pa_context_errno(c->context),
>> + "set_source_output_mute() failed\n");
>> } else {
>> pa_operation_unref (op);
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> }
>> }
>> return 0;
>> @@ -828,11 +840,75 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
>> return 1;
>> }
>>
>> +/* common */
>> +static void *qpa_conn_init(const char *server)
>> +{
>> + PAConnection *c = g_malloc0(sizeof(PAConnection));
>> + QTAILQ_INSERT_TAIL(&pa_conns, c, list);
>> +
>> + c->mainloop = pa_threaded_mainloop_new();
>> + if (!c->mainloop) {
>> + goto fail;
>> + }
>> +
>> + c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
>> + server);
>> + if (!c->context) {
>> + goto fail;
>> + }
>> +
>> + pa_context_set_state_callback(c->context, context_state_cb, c);
>> +
>> + if (pa_context_connect(c->context, server, 0, NULL) < 0) {
>> + qpa_logerr(pa_context_errno(c->context),
>> + "pa_context_connect() failed\n");
>> + goto fail;
>> + }
>> +
>> + pa_threaded_mainloop_lock(c->mainloop);
>> +
>> + if (pa_threaded_mainloop_start(c->mainloop) < 0) {
>> + goto unlock_and_fail;
>> + }
>> +
>> + for (;;) {
>> + pa_context_state_t state;
>> +
>> + state = pa_context_get_state(c->context);
>> +
>> + if (state == PA_CONTEXT_READY) {
>> + break;
>> + }
>> +
>> + if (!PA_CONTEXT_IS_GOOD(state)) {
>> + qpa_logerr(pa_context_errno(c->context),
>> + "Wrong context state\n");
>> + goto unlock_and_fail;
>> + }
>> +
>> + /* Wait until the context is ready */
>> + pa_threaded_mainloop_wait(c->mainloop);
>> + }
>> +
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> + return c;
>> +
>> +unlock_and_fail:
>> + pa_threaded_mainloop_unlock(c->mainloop);
>> +fail:
>> + AUD_log (AUDIO_CAP, "Failed to initialize PA context");
>> + qpa_conn_fini(c);
>> + return NULL;
>> +}
>> +
>> static void *qpa_audio_init(Audiodev *dev)
>> {
>> paaudio *g;
>> AudiodevPaOptions *popts = &dev->u.pa;
>> const char *server;
>> + PAConnection *c;
>> +
>> + assert(dev->driver == AUDIODEV_DRIVER_PA);
>>
>> if (!popts->has_server) {
>> char pidfile[64];
>> @@ -849,93 +925,64 @@ static void *qpa_audio_init(Audiodev *dev)
>> }
>> }
>>
>> - assert(dev->driver == AUDIODEV_DRIVER_PA);
>> -
>> - g = g_malloc(sizeof(paaudio));
>> - server = popts->has_server ? popts->server : NULL;
>> -
>> if (!qpa_validate_per_direction_opts(dev, popts->in)) {
>> - goto fail;
>> + return NULL;
>> }
>> if (!qpa_validate_per_direction_opts(dev, popts->out)) {
>> - goto fail;
>> + return NULL;
>> }
>>
>> + g = g_malloc0(sizeof(paaudio));
>> + server = popts->has_server ? popts->server : NULL;
>> +
>> g->dev = dev;
>> - g->mainloop = NULL;
>> - g->context = NULL;
>>
>> - g->mainloop = pa_threaded_mainloop_new ();
>> - if (!g->mainloop) {
>> - goto fail;
>> - }
>> -
>> - g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
>> - server);
>> - if (!g->context) {
>> - goto fail;
>> - }
>> -
>> - pa_context_set_state_callback (g->context, context_state_cb, g);
>> -
>> - if (pa_context_connect(g->context, server, 0, NULL) < 0) {
>> - qpa_logerr (pa_context_errno (g->context),
>> - "pa_context_connect() failed\n");
>> - goto fail;
>> - }
>> -
>> - pa_threaded_mainloop_lock (g->mainloop);
>> -
>> - if (pa_threaded_mainloop_start (g->mainloop) < 0) {
>> - goto unlock_and_fail;
>> - }
>> -
>> - for (;;) {
>> - pa_context_state_t state;
>> -
>> - state = pa_context_get_state (g->context);
>> -
>> - if (state == PA_CONTEXT_READY) {
>> + QTAILQ_FOREACH(c, &pa_conns, list) {
>> + if (server == NULL || c->server == NULL ?
>> + server == c->server :
>> + strcmp(server, c->server) == 0) {
>> + g->conn = c;
>> break;
>> }
>> -
>> - if (!PA_CONTEXT_IS_GOOD (state)) {
>> - qpa_logerr (pa_context_errno (g->context),
>> - "Wrong context state\n");
>> - goto unlock_and_fail;
>> - }
>> -
>> - /* Wait until the context is ready */
>> - pa_threaded_mainloop_wait (g->mainloop);
>> + }
>> + if (!g->conn) {
>> + g->conn = qpa_conn_init(server);
>> + }
>> + if (!g->conn) {
>> + g_free(g);
>> + return NULL;
>> }
>>
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> -
>> + ++g->conn->refcount;
>> return g;
>> +}
>>
>> -unlock_and_fail:
>> - pa_threaded_mainloop_unlock (g->mainloop);
>> -fail:
>> - AUD_log (AUDIO_CAP, "Failed to initialize PA context");
>> - qpa_audio_fini(g);
>> - return NULL;
>> +static void qpa_conn_fini(PAConnection *c)
>> +{
>> + if (c->mainloop) {
>> + pa_threaded_mainloop_stop(c->mainloop);
>> + }
>> +
>> + if (c->context) {
>> + pa_context_disconnect(c->context);
>> + pa_context_unref(c->context);
>> + }
>> +
>> + if (c->mainloop) {
>> + pa_threaded_mainloop_free(c->mainloop);
>> + }
>> +
>> + QTAILQ_REMOVE(&pa_conns, c, list);
>> + g_free(c);
>> }
>>
>> static void qpa_audio_fini (void *opaque)
>> {
>> paaudio *g = opaque;
>> + PAConnection *c = g->conn;
>>
>> - if (g->mainloop) {
>> - pa_threaded_mainloop_stop (g->mainloop);
>> - }
>> -
>> - if (g->context) {
>> - pa_context_disconnect (g->context);
>> - pa_context_unref (g->context);
>> - }
>> -
>> - if (g->mainloop) {
>> - pa_threaded_mainloop_free (g->mainloop);
>> + if (--c->refcount == 0) {
>> + qpa_conn_fini(c);
>> }
>>
>> g_free(g);
>> --
>> 2.22.0
>>
>>
>
>
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 05/11] paaudio: do not move stream when sink/source name is specified
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (3 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 04/11] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 06/11] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
` (7 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Unless we disable stream moving, pulseaudio can easily move the stream
on connect, effectively ignoring the source/sink specified by the user.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index cc3a34c2ea..24d98b344a 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -517,6 +517,11 @@ static pa_stream *qpa_simple_new (
#endif
| PA_STREAM_AUTO_TIMING_UPDATE;
+ if (dev) {
+ /* don't move the stream if the user specified a sink/source */
+ flags |= PA_STREAM_DONT_MOVE;
+ }
+
if (dir == PA_STREAM_PLAYBACK) {
r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
} else {
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 05/11] paaudio: do not move stream when sink/source name is specified
2019-07-09 18:35 ` [Qemu-devel] [PATCH 05/11] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:53 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Unless we disable stream moving, pulseaudio can easily move the stream
> on connect, effectively ignoring the source/sink specified by the user.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> audio/paaudio.c | 5 +++++
> 1 file changed, 5 insertions(+)
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index cc3a34c2ea..24d98b344a 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -517,6 +517,11 @@ static pa_stream *qpa_simple_new (
> #endif
> | PA_STREAM_AUTO_TIMING_UPDATE;
>
> + if (dev) {
> + /* don't move the stream if the user specified a sink/source */
> + flags |= PA_STREAM_DONT_MOVE;
> + }
> +
> if (dir == PA_STREAM_PLAYBACK) {
> r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
> } else {
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 06/11] paaudio: properly disconnect streams in fini_*
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (4 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 05/11] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
` (6 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Currently this needs a workaround due to bug #247 in pulseaudio.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 25 +++++++++++++++++++++++--
1 file changed, 23 insertions(+), 2 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 24d98b344a..490bcd770e 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -685,6 +685,27 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return -1;
}
+static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
+{
+ int err;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+ /*
+ * wait until actually connects. workaround pa bug #247
+ * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/247
+ */
+ while (pa_stream_get_state(stream) == PA_STREAM_CREATING) {
+ pa_threaded_mainloop_wait(c->mainloop);
+ }
+
+ err = pa_stream_disconnect(stream);
+ if (err != 0) {
+ dolog("Failed to dissconnect! err=%d\n", err);
+ }
+ pa_stream_unref(stream);
+ pa_threaded_mainloop_unlock(c->mainloop);
+}
+
static void qpa_fini_out (HWVoiceOut *hw)
{
void *ret;
@@ -696,7 +717,7 @@ static void qpa_fini_out (HWVoiceOut *hw)
audio_pt_join(&pa->pt, &ret, __func__);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
@@ -716,7 +737,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
audio_pt_join(&pa->pt, &ret, __func__);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 06/11] paaudio: properly disconnect streams in fini_*
2019-07-09 18:35 ` [Qemu-devel] [PATCH 06/11] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:53 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Currently this needs a workaround due to bug #247 in pulseaudio.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> audio/paaudio.c | 25 +++++++++++++++++++++++--
> 1 file changed, 23 insertions(+), 2 deletions(-)
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index 24d98b344a..490bcd770e 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -685,6 +685,27 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> return -1;
> }
>
> +static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
> +{
> + int err;
> +
> + pa_threaded_mainloop_lock(c->mainloop);
> + /*
> + * wait until actually connects. workaround pa bug #247
> + * https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/247
> + */
> + while (pa_stream_get_state(stream) == PA_STREAM_CREATING) {
> + pa_threaded_mainloop_wait(c->mainloop);
> + }
> +
> + err = pa_stream_disconnect(stream);
> + if (err != 0) {
> + dolog("Failed to dissconnect! err=%d\n", err);
disconnect
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> + }
> + pa_stream_unref(stream);
> + pa_threaded_mainloop_unlock(c->mainloop);
> +}
> +
> static void qpa_fini_out (HWVoiceOut *hw)
> {
> void *ret;
> @@ -696,7 +717,7 @@ static void qpa_fini_out (HWVoiceOut *hw)
> audio_pt_join(&pa->pt, &ret, __func__);
>
> if (pa->stream) {
> - pa_stream_unref (pa->stream);
> + qpa_simple_disconnect(pa->g->conn, pa->stream);
> pa->stream = NULL;
> }
>
> @@ -716,7 +737,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
> audio_pt_join(&pa->pt, &ret, __func__);
>
> if (pa->stream) {
> - pa_stream_unref (pa->stream);
> + qpa_simple_disconnect(pa->g->conn, pa->stream);
> pa->stream = NULL;
> }
>
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (5 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 06/11] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 08/11] audio: do not run each backend in audio_run Kővágó, Zoltán
` (5 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Michael Walle, Gerd Hoffmann
There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
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
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX
2019-07-09 18:35 ` [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: Michael Walle, QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 11:08 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
> instead.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> 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
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 08/11] audio: do not run each backend in audio_run
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (6 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 07/11] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches Kővágó, Zoltán
` (4 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
audio_run is called manually by alsa and oss backends when polling.
In this case only the requesting backend should be run, not all of them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_int.h | 2 +-
audio/alsaaudio.c | 7 +++++--
audio/audio.c | 14 +++++---------
audio/ossaudio.c | 12 ++++++------
4 files changed, 17 insertions(+), 18 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9f01f6ad00..7e00c1332e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -221,7 +221,7 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
-void audio_run (const char *msg);
+void audio_run(AudioState *s, const char *msg);
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 6b9e0f06af..3daa7c8f8f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -39,6 +39,7 @@ struct pollhlp {
struct pollfd *pfds;
int count;
int mask;
+ AudioState *s;
};
typedef struct ALSAVoiceOut {
@@ -199,11 +200,11 @@ static void alsa_poll_handler (void *opaque)
break;
case SND_PCM_STATE_PREPARED:
- audio_run ("alsa run (prepared)");
+ audio_run(hlp->s, "alsa run (prepared)");
break;
case SND_PCM_STATE_RUNNING:
- audio_run ("alsa run (running)");
+ audio_run(hlp->s, "alsa run (running)");
break;
default:
@@ -749,6 +750,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
+ alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
@@ -850,6 +852,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return -1;
}
+ alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
diff --git a/audio/audio.c b/audio/audio.c
index 6bf30ac9b3..d73cc086b6 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -835,7 +835,7 @@ static void audio_timer (void *opaque)
}
s->timer_last = now;
- audio_run("timer");
+ audio_run(s, "timer");
audio_reset_timer(s);
}
@@ -1237,15 +1237,11 @@ static void audio_run_capture (AudioState *s)
}
}
-void audio_run (const char *msg)
+void audio_run(AudioState *s, const char *msg)
{
- AudioState *s;
-
- QTAILQ_FOREACH(s, &audio_states, list) {
- audio_run_out(s);
- audio_run_in(s);
- audio_run_capture(s);
- }
+ audio_run_out(s);
+ audio_run_in(s);
+ audio_run_capture(s);
#ifdef DEBUG_POLL
{
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 29139ef1f5..456fba7480 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -110,28 +110,28 @@ static void oss_anal_close (int *fdp)
static void oss_helper_poll_out (void *opaque)
{
- (void) opaque;
- audio_run ("oss_poll_out");
+ AudioState *s = opaque;
+ audio_run(s, "oss_poll_out");
}
static void oss_helper_poll_in (void *opaque)
{
- (void) opaque;
- audio_run ("oss_poll_in");
+ AudioState *s = opaque;
+ audio_run(s, "oss_poll_in");
}
static void oss_poll_out (HWVoiceOut *hw)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- qemu_set_fd_handler (oss->fd, NULL, oss_helper_poll_out, NULL);
+ qemu_set_fd_handler(oss->fd, NULL, oss_helper_poll_out, hw->s);
}
static void oss_poll_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
- qemu_set_fd_handler (oss->fd, oss_helper_poll_in, NULL, NULL);
+ qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
}
static int oss_write (SWVoiceOut *sw, void *buf, int len)
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 08/11] audio: do not run each backend in audio_run
2019-07-09 18:35 ` [Qemu-devel] [PATCH 08/11] audio: do not run each backend in audio_run Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
0 siblings, 0 replies; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:51 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> audio_run is called manually by alsa and oss backends when polling.
> In this case only the requesting backend should be run, not all of them.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
> ---
> audio/audio_int.h | 2 +-
> audio/alsaaudio.c | 7 +++++--
> audio/audio.c | 14 +++++---------
> audio/ossaudio.c | 12 ++++++------
> 4 files changed, 17 insertions(+), 18 deletions(-)
>
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index 9f01f6ad00..7e00c1332e 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -221,7 +221,7 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> int audio_bug (const char *funcname, int cond);
> void *audio_calloc (const char *funcname, int nmemb, size_t size);
>
> -void audio_run (const char *msg);
> +void audio_run(AudioState *s, const char *msg);
>
> #define VOICE_ENABLE 1
> #define VOICE_DISABLE 2
> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
> index 6b9e0f06af..3daa7c8f8f 100644
> --- a/audio/alsaaudio.c
> +++ b/audio/alsaaudio.c
> @@ -39,6 +39,7 @@ struct pollhlp {
> struct pollfd *pfds;
> int count;
> int mask;
> + AudioState *s;
> };
>
> typedef struct ALSAVoiceOut {
> @@ -199,11 +200,11 @@ static void alsa_poll_handler (void *opaque)
> break;
>
> case SND_PCM_STATE_PREPARED:
> - audio_run ("alsa run (prepared)");
> + audio_run(hlp->s, "alsa run (prepared)");
> break;
>
> case SND_PCM_STATE_RUNNING:
> - audio_run ("alsa run (running)");
> + audio_run(hlp->s, "alsa run (running)");
> break;
>
> default:
> @@ -749,6 +750,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
> return -1;
> }
>
> + alsa->pollhlp.s = hw->s;
> alsa->handle = handle;
> alsa->dev = dev;
> return 0;
> @@ -850,6 +852,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> return -1;
> }
>
> + alsa->pollhlp.s = hw->s;
> alsa->handle = handle;
> alsa->dev = dev;
> return 0;
> diff --git a/audio/audio.c b/audio/audio.c
> index 6bf30ac9b3..d73cc086b6 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -835,7 +835,7 @@ static void audio_timer (void *opaque)
> }
> s->timer_last = now;
>
> - audio_run("timer");
> + audio_run(s, "timer");
> audio_reset_timer(s);
> }
>
> @@ -1237,15 +1237,11 @@ static void audio_run_capture (AudioState *s)
> }
> }
>
> -void audio_run (const char *msg)
> +void audio_run(AudioState *s, const char *msg)
> {
> - AudioState *s;
> -
> - QTAILQ_FOREACH(s, &audio_states, list) {
> - audio_run_out(s);
> - audio_run_in(s);
> - audio_run_capture(s);
> - }
> + audio_run_out(s);
> + audio_run_in(s);
> + audio_run_capture(s);
>
> #ifdef DEBUG_POLL
> {
> diff --git a/audio/ossaudio.c b/audio/ossaudio.c
> index 29139ef1f5..456fba7480 100644
> --- a/audio/ossaudio.c
> +++ b/audio/ossaudio.c
> @@ -110,28 +110,28 @@ static void oss_anal_close (int *fdp)
>
> static void oss_helper_poll_out (void *opaque)
> {
> - (void) opaque;
> - audio_run ("oss_poll_out");
> + AudioState *s = opaque;
> + audio_run(s, "oss_poll_out");
> }
>
> static void oss_helper_poll_in (void *opaque)
> {
> - (void) opaque;
> - audio_run ("oss_poll_in");
> + AudioState *s = opaque;
> + audio_run(s, "oss_poll_in");
> }
>
> static void oss_poll_out (HWVoiceOut *hw)
> {
> OSSVoiceOut *oss = (OSSVoiceOut *) hw;
>
> - qemu_set_fd_handler (oss->fd, NULL, oss_helper_poll_out, NULL);
> + qemu_set_fd_handler(oss->fd, NULL, oss_helper_poll_out, hw->s);
> }
>
> static void oss_poll_in (HWVoiceIn *hw)
> {
> OSSVoiceIn *oss = (OSSVoiceIn *) hw;
>
> - qemu_set_fd_handler (oss->fd, oss_helper_poll_in, NULL, NULL);
> + qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
> }
>
> static int oss_write (SWVoiceOut *sw, void *buf, int len)
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (7 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 08/11] audio: do not run each backend in audio_run Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:58 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops Kővágó, Zoltán
` (3 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Pulseaudio normally assumes that when the server wants it, the client
can generate the audio samples and send it right away. Unfortunately
this is not the case with QEMU -- it's up to the emulated system when
does it generate the samples. Buffering the samples and sending them
from a background thread is just a workaround, that doesn't work too
well. Instead enable pa's compatibility support and let pa worry about
the details.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 6 ++----
1 file changed, 2 insertions(+), 4 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 9d46f11b0a..d320d2e453 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -512,10 +512,8 @@ static pa_stream *qpa_simple_new (
flags =
PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- | PA_STREAM_ADJUST_LATENCY
-#endif
- | PA_STREAM_AUTO_TIMING_UPDATE;
+ | PA_STREAM_AUTO_TIMING_UPDATE
+ | PA_STREAM_EARLY_REQUESTS;
if (dev) {
/* don't move the stream if the user specified a sink/source */
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches
2019-07-09 18:35 ` [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches Kővágó, Zoltán
@ 2019-07-10 19:58 ` Marc-André Lureau
2019-07-14 15:21 ` Zoltán Kővágó
0 siblings, 1 reply; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:58 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:49 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> Pulseaudio normally assumes that when the server wants it, the client
> can generate the audio samples and send it right away. Unfortunately
> this is not the case with QEMU -- it's up to the emulated system when
> does it generate the samples. Buffering the samples and sending them
> from a background thread is just a workaround, that doesn't work too
> well. Instead enable pa's compatibility support and let pa worry about
> the details.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Could you explain how this is related to PA_STREAM_ADJUST_LATENCY ?
> ---
> audio/paaudio.c | 6 ++----
> 1 file changed, 2 insertions(+), 4 deletions(-)
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index 9d46f11b0a..d320d2e453 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -512,10 +512,8 @@ static pa_stream *qpa_simple_new (
>
> flags =
> PA_STREAM_INTERPOLATE_TIMING
> -#ifdef PA_STREAM_ADJUST_LATENCY
> - | PA_STREAM_ADJUST_LATENCY
> -#endif
> - | PA_STREAM_AUTO_TIMING_UPDATE;
> + | PA_STREAM_AUTO_TIMING_UPDATE
> + | PA_STREAM_EARLY_REQUESTS;
>
> if (dev) {
> /* don't move the stream if the user specified a sink/source */
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches
2019-07-10 19:58 ` Marc-André Lureau
@ 2019-07-14 15:21 ` Zoltán Kővágó
0 siblings, 0 replies; 32+ messages in thread
From: Zoltán Kővágó @ 2019-07-14 15:21 UTC (permalink / raw)
To: Marc-André Lureau; +Cc: QEMU, Gerd Hoffmann
On 2019-07-10 21:58, Marc-André Lureau wrote:
> On Tue, Jul 9, 2019 at 10:49 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>>
>> Pulseaudio normally assumes that when the server wants it, the client
>> can generate the audio samples and send it right away. Unfortunately
>> this is not the case with QEMU -- it's up to the emulated system when
>> does it generate the samples. Buffering the samples and sending them
>> from a background thread is just a workaround, that doesn't work too
>> well. Instead enable pa's compatibility support and let pa worry about
>> the details.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>
> Could you explain how this is related to PA_STREAM_ADJUST_LATENCY ?
According to the pulseaudio documentation[1], you can't use
PA_STREAM_ADJUST_LATENCY and PA_STREAM_EARLY_REQUESTS at the same time.
[1]:
https://www.freedesktop.org/software/pulseaudio/doxygen/def_8h.html#a6966d809483170bc6d2e6c16188850fca98e436f686fc385697e565eb1ecb2609
>> ---
>> audio/paaudio.c | 6 ++----
>> 1 file changed, 2 insertions(+), 4 deletions(-)
>>
>> diff --git a/audio/paaudio.c b/audio/paaudio.c
>> index 9d46f11b0a..d320d2e453 100644
>> --- a/audio/paaudio.c
>> +++ b/audio/paaudio.c
>> @@ -512,10 +512,8 @@ static pa_stream *qpa_simple_new (
>>
>> flags =
>> PA_STREAM_INTERPOLATE_TIMING
>> -#ifdef PA_STREAM_ADJUST_LATENCY
>> - | PA_STREAM_ADJUST_LATENCY
>> -#endif
>> - | PA_STREAM_AUTO_TIMING_UPDATE;
>> + | PA_STREAM_AUTO_TIMING_UPDATE
>> + | PA_STREAM_EARLY_REQUESTS;
>>
>> if (dev) {
>> /* don't move the stream if the user specified a sink/source */
>> --
>> 2.22.0
>>
>>
>
>
> --
> Marc-André Lureau
>
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (8 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 09/11] paaudio: fix playback glitches Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-10 19:57 ` Marc-André Lureau
2019-07-09 18:35 ` [Qemu-devel] [PATCH 11/11] audio: use size_t where makes sense Kővágó, Zoltán
` (2 subsequent siblings)
12 siblings, 1 reply; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
They just called audio_pcm_sw_read/write anyway, so it makes no sense
to have them too. (The noaudio's read is the only exception, but it
should work with the generic code too.)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_int.h | 5 -----
audio/alsaaudio.c | 12 ------------
audio/audio.c | 8 ++++----
audio/coreaudio.c | 6 ------
audio/dsoundaudio.c | 12 ------------
audio/noaudio.c | 19 -------------------
audio/ossaudio.c | 12 ------------
audio/paaudio.c | 12 ------------
audio/sdlaudio.c | 6 ------
audio/spiceaudio.c | 12 ------------
audio/wavaudio.c | 6 ------
11 files changed, 4 insertions(+), 106 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 7e00c1332e..003b7ab8cc 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -150,13 +150,11 @@ struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
int (*run_out) (HWVoiceOut *hw, int live);
- int (*write) (SWVoiceOut *sw, void *buf, int size);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
int (*run_in) (HWVoiceIn *hw);
- int (*read) (SWVoiceIn *sw, void *buf, int size);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
@@ -210,11 +208,8 @@ 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_sw_write (SWVoiceOut *sw, void *buf, int len);
int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
-int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
-
int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int live, int pending);
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 3daa7c8f8f..e9e3a4819c 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -270,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw)
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}
-static int alsa_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
@@ -988,11 +983,6 @@ static int alsa_run_in (HWVoiceIn *hw)
return read_samples;
}
-static int alsa_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
@@ -1076,13 +1066,11 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.run_out = alsa_run_out,
- .write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.run_in = alsa_run_in,
- .read = alsa_read,
.ctl_in = alsa_ctl_in,
};
diff --git a/audio/audio.c b/audio/audio.c
index d73cc086b6..b79f56fe64 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -594,7 +594,7 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
}
}
-int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
+static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
{
HWVoiceIn *hw = sw->hw;
int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
@@ -696,7 +696,7 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
/*
* Soft voice (playback)
*/
-int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
+static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf, int size)
{
int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
int ret = 0, pos = 0, total = 0;
@@ -854,7 +854,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
return 0;
}
- return sw->hw->pcm_ops->write(sw, buf, size);
+ return audio_pcm_sw_write(sw, buf, size);
}
int AUD_read (SWVoiceIn *sw, void *buf, int size)
@@ -869,7 +869,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size)
return 0;
}
- return sw->hw->pcm_ops->read(sw, buf, size);
+ return audio_pcm_sw_read(sw, buf, size);
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index f0ab4014a8..091fe84a34 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -489,11 +489,6 @@ static OSStatus audioDeviceIOProc(
return 0;
}
-static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
@@ -692,7 +687,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
.run_out = coreaudio_run_out,
- .write = coreaudio_write,
.ctl_out = coreaudio_ctl_out
};
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 07260f881e..11594c3095 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -454,11 +454,6 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int dsound_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int dsound_run_out (HWVoiceOut *hw, int live)
{
int err;
@@ -645,11 +640,6 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static int dsound_read (SWVoiceIn *sw, void *buf, int len)
-{
- return audio_pcm_sw_read (sw, buf, len);
-}
-
static int dsound_run_in (HWVoiceIn *hw)
{
int err;
@@ -856,13 +846,11 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
.run_out = dsound_run_out,
- .write = dsound_write,
.ctl_out = dsound_ctl_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
.run_in = dsound_run_in,
- .read = dsound_read,
.ctl_in = dsound_ctl_in
};
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 14a0e4ab29..cbb02d9e49 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -61,11 +61,6 @@ static int no_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int no_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write(sw, buf, len);
-}
-
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
@@ -118,18 +113,6 @@ static int no_run_in (HWVoiceIn *hw)
return samples;
}
-static int no_read (SWVoiceIn *sw, void *buf, int size)
-{
- /* use custom code here instead of audio_pcm_sw_read() to avoid
- * useless resampling/mixing */
- int samples = size >> sw->info.shift;
- int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- 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;
-}
-
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
@@ -151,13 +134,11 @@ static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
.run_out = no_run_out,
- .write = no_write,
.ctl_out = no_ctl_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
.run_in = no_run_in,
- .read = no_read,
.ctl_in = no_ctl_in
};
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 456fba7480..b99edbec17 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -134,11 +134,6 @@ static void oss_poll_in (HWVoiceIn *hw)
qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
}
-static int oss_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
@@ -788,11 +783,6 @@ static int oss_run_in (HWVoiceIn *hw)
return read_samples;
}
-static int oss_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
@@ -855,13 +845,11 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.run_out = oss_run_out,
- .write = oss_write,
.ctl_out = oss_ctl_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.run_in = oss_run_in,
- .read = oss_read,
.ctl_in = oss_ctl_in
};
diff --git a/audio/paaudio.c b/audio/paaudio.c
index d320d2e453..12471154ed 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -295,11 +295,6 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int qpa_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
/* capture */
static void *qpa_thread_in (void *arg)
{
@@ -388,11 +383,6 @@ static int qpa_run_in (HWVoiceIn *hw)
return incr;
}
-static int qpa_read (SWVoiceIn *sw, void *buf, int len)
-{
- return audio_pcm_sw_read (sw, buf, len);
-}
-
static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
{
int format;
@@ -1016,13 +1006,11 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.run_out = qpa_run_out,
- .write = qpa_write,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.run_in = qpa_run_in,
- .read = qpa_read,
.ctl_in = qpa_ctl_in
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 42f7614124..c7fd487e0e 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -217,11 +217,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
}
-static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int sdl_run_out (HWVoiceOut *hw, int live)
{
int decr;
@@ -342,7 +337,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
.run_out = sdl_run_out,
- .write = sdl_write_out,
.ctl_out = sdl_ctl_out,
};
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6f4a0558f8..6e7f13cffa 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -193,11 +193,6 @@ static int line_out_run (HWVoiceOut *hw, int live)
return decr;
}
-static int line_out_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
@@ -326,11 +321,6 @@ static int line_in_run (HWVoiceIn *hw)
return num_samples;
}
-static int line_in_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
@@ -378,13 +368,11 @@ static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
.run_out = line_out_run,
- .write = line_out_write,
.ctl_out = line_out_ctl,
.init_in = line_in_init,
.fini_in = line_in_fini,
.run_in = line_in_run,
- .read = line_in_read,
.ctl_in = line_in_ctl,
};
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index bbf3f3b346..17ab921cef 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -84,11 +84,6 @@ static int wav_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
/* VICE code: Store number as little endian. */
static void le_store (uint8_t *buf, uint32_t val, int len)
{
@@ -240,7 +235,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.run_out = wav_run_out,
- .write = wav_write_out,
.ctl_out = wav_ctl_out,
};
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops
2019-07-09 18:35 ` [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops Kővágó, Zoltán
@ 2019-07-10 19:57 ` Marc-André Lureau
2019-07-11 19:20 ` Zoltán Kővágó
0 siblings, 1 reply; 32+ messages in thread
From: Marc-André Lureau @ 2019-07-10 19:57 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: QEMU, Gerd Hoffmann
On Tue, Jul 9, 2019 at 10:57 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>
> They just called audio_pcm_sw_read/write anyway, so it makes no sense
> to have them too. (The noaudio's read is the only exception, but it
> should work with the generic code too.)
It works with the generic code, but wouldn't it be more expensive?
Perhaps there can be something in audio_pcm_sw_write() to skip the
work if noaudio is the backend?
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> audio/audio_int.h | 5 -----
> audio/alsaaudio.c | 12 ------------
> audio/audio.c | 8 ++++----
> audio/coreaudio.c | 6 ------
> audio/dsoundaudio.c | 12 ------------
> audio/noaudio.c | 19 -------------------
> audio/ossaudio.c | 12 ------------
> audio/paaudio.c | 12 ------------
> audio/sdlaudio.c | 6 ------
> audio/spiceaudio.c | 12 ------------
> audio/wavaudio.c | 6 ------
> 11 files changed, 4 insertions(+), 106 deletions(-)
>
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index 7e00c1332e..003b7ab8cc 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -150,13 +150,11 @@ struct audio_pcm_ops {
> int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
> void (*fini_out)(HWVoiceOut *hw);
> int (*run_out) (HWVoiceOut *hw, int live);
> - int (*write) (SWVoiceOut *sw, void *buf, int size);
> int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
>
> int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
> void (*fini_in) (HWVoiceIn *hw);
> int (*run_in) (HWVoiceIn *hw);
> - int (*read) (SWVoiceIn *sw, void *buf, int size);
> int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
> };
>
> @@ -210,11 +208,8 @@ 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_sw_write (SWVoiceOut *sw, void *buf, int len);
> int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
>
> -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
> -
> int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> int live, int pending);
>
> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
> index 3daa7c8f8f..e9e3a4819c 100644
> --- a/audio/alsaaudio.c
> +++ b/audio/alsaaudio.c
> @@ -270,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw)
> return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
> }
>
> -static int alsa_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
> {
> switch (fmt) {
> @@ -988,11 +983,6 @@ static int alsa_run_in (HWVoiceIn *hw)
> return read_samples;
> }
>
> -static int alsa_read (SWVoiceIn *sw, void *buf, int size)
> -{
> - return audio_pcm_sw_read (sw, buf, size);
> -}
> -
> static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
> {
> ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
> @@ -1076,13 +1066,11 @@ static struct audio_pcm_ops alsa_pcm_ops = {
> .init_out = alsa_init_out,
> .fini_out = alsa_fini_out,
> .run_out = alsa_run_out,
> - .write = alsa_write,
> .ctl_out = alsa_ctl_out,
>
> .init_in = alsa_init_in,
> .fini_in = alsa_fini_in,
> .run_in = alsa_run_in,
> - .read = alsa_read,
> .ctl_in = alsa_ctl_in,
> };
>
> diff --git a/audio/audio.c b/audio/audio.c
> index d73cc086b6..b79f56fe64 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -594,7 +594,7 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
> }
> }
>
> -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
> +static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
> {
> HWVoiceIn *hw = sw->hw;
> int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
> @@ -696,7 +696,7 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> /*
> * Soft voice (playback)
> */
> -int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
> +static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf, int size)
> {
> int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
> int ret = 0, pos = 0, total = 0;
> @@ -854,7 +854,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
> return 0;
> }
>
> - return sw->hw->pcm_ops->write(sw, buf, size);
> + return audio_pcm_sw_write(sw, buf, size);
> }
>
> int AUD_read (SWVoiceIn *sw, void *buf, int size)
> @@ -869,7 +869,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size)
> return 0;
> }
>
> - return sw->hw->pcm_ops->read(sw, buf, size);
> + return audio_pcm_sw_read(sw, buf, size);
> }
>
> int AUD_get_buffer_size_out (SWVoiceOut *sw)
> diff --git a/audio/coreaudio.c b/audio/coreaudio.c
> index f0ab4014a8..091fe84a34 100644
> --- a/audio/coreaudio.c
> +++ b/audio/coreaudio.c
> @@ -489,11 +489,6 @@ static OSStatus audioDeviceIOProc(
> return 0;
> }
>
> -static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
> void *drv_opaque)
> {
> @@ -692,7 +687,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
> .init_out = coreaudio_init_out,
> .fini_out = coreaudio_fini_out,
> .run_out = coreaudio_run_out,
> - .write = coreaudio_write,
> .ctl_out = coreaudio_ctl_out
> };
>
> diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
> index 07260f881e..11594c3095 100644
> --- a/audio/dsoundaudio.c
> +++ b/audio/dsoundaudio.c
> @@ -454,11 +454,6 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
> return 0;
> }
>
> -static int dsound_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static int dsound_run_out (HWVoiceOut *hw, int live)
> {
> int err;
> @@ -645,11 +640,6 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
> return 0;
> }
>
> -static int dsound_read (SWVoiceIn *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_read (sw, buf, len);
> -}
> -
> static int dsound_run_in (HWVoiceIn *hw)
> {
> int err;
> @@ -856,13 +846,11 @@ static struct audio_pcm_ops dsound_pcm_ops = {
> .init_out = dsound_init_out,
> .fini_out = dsound_fini_out,
> .run_out = dsound_run_out,
> - .write = dsound_write,
> .ctl_out = dsound_ctl_out,
>
> .init_in = dsound_init_in,
> .fini_in = dsound_fini_in,
> .run_in = dsound_run_in,
> - .read = dsound_read,
> .ctl_in = dsound_ctl_in
> };
>
> diff --git a/audio/noaudio.c b/audio/noaudio.c
> index 14a0e4ab29..cbb02d9e49 100644
> --- a/audio/noaudio.c
> +++ b/audio/noaudio.c
> @@ -61,11 +61,6 @@ static int no_run_out (HWVoiceOut *hw, int live)
> return decr;
> }
>
> -static int no_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write(sw, buf, len);
> -}
> -
> static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
> {
> audio_pcm_init_info (&hw->info, as);
> @@ -118,18 +113,6 @@ static int no_run_in (HWVoiceIn *hw)
> return samples;
> }
>
> -static int no_read (SWVoiceIn *sw, void *buf, int size)
> -{
> - /* use custom code here instead of audio_pcm_sw_read() to avoid
> - * useless resampling/mixing */
> - int samples = size >> sw->info.shift;
> - int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
> - 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;
> -}
> -
> static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
> {
> (void) hw;
> @@ -151,13 +134,11 @@ static struct audio_pcm_ops no_pcm_ops = {
> .init_out = no_init_out,
> .fini_out = no_fini_out,
> .run_out = no_run_out,
> - .write = no_write,
> .ctl_out = no_ctl_out,
>
> .init_in = no_init_in,
> .fini_in = no_fini_in,
> .run_in = no_run_in,
> - .read = no_read,
> .ctl_in = no_ctl_in
> };
>
> diff --git a/audio/ossaudio.c b/audio/ossaudio.c
> index 456fba7480..b99edbec17 100644
> --- a/audio/ossaudio.c
> +++ b/audio/ossaudio.c
> @@ -134,11 +134,6 @@ static void oss_poll_in (HWVoiceIn *hw)
> qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
> }
>
> -static int oss_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static int aud_to_ossfmt (AudioFormat fmt, int endianness)
> {
> switch (fmt) {
> @@ -788,11 +783,6 @@ static int oss_run_in (HWVoiceIn *hw)
> return read_samples;
> }
>
> -static int oss_read (SWVoiceIn *sw, void *buf, int size)
> -{
> - return audio_pcm_sw_read (sw, buf, size);
> -}
> -
> static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
> {
> OSSVoiceIn *oss = (OSSVoiceIn *) hw;
> @@ -855,13 +845,11 @@ static struct audio_pcm_ops oss_pcm_ops = {
> .init_out = oss_init_out,
> .fini_out = oss_fini_out,
> .run_out = oss_run_out,
> - .write = oss_write,
> .ctl_out = oss_ctl_out,
>
> .init_in = oss_init_in,
> .fini_in = oss_fini_in,
> .run_in = oss_run_in,
> - .read = oss_read,
> .ctl_in = oss_ctl_in
> };
>
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index d320d2e453..12471154ed 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -295,11 +295,6 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
> return decr;
> }
>
> -static int qpa_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> /* capture */
> static void *qpa_thread_in (void *arg)
> {
> @@ -388,11 +383,6 @@ static int qpa_run_in (HWVoiceIn *hw)
> return incr;
> }
>
> -static int qpa_read (SWVoiceIn *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_read (sw, buf, len);
> -}
> -
> static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
> {
> int format;
> @@ -1016,13 +1006,11 @@ static struct audio_pcm_ops qpa_pcm_ops = {
> .init_out = qpa_init_out,
> .fini_out = qpa_fini_out,
> .run_out = qpa_run_out,
> - .write = qpa_write,
> .ctl_out = qpa_ctl_out,
>
> .init_in = qpa_init_in,
> .fini_in = qpa_fini_in,
> .run_in = qpa_run_in,
> - .read = qpa_read,
> .ctl_in = qpa_ctl_in
> };
>
> diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
> index 42f7614124..c7fd487e0e 100644
> --- a/audio/sdlaudio.c
> +++ b/audio/sdlaudio.c
> @@ -217,11 +217,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
> }
> }
>
> -static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static int sdl_run_out (HWVoiceOut *hw, int live)
> {
> int decr;
> @@ -342,7 +337,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
> .init_out = sdl_init_out,
> .fini_out = sdl_fini_out,
> .run_out = sdl_run_out,
> - .write = sdl_write_out,
> .ctl_out = sdl_ctl_out,
> };
>
> diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
> index 6f4a0558f8..6e7f13cffa 100644
> --- a/audio/spiceaudio.c
> +++ b/audio/spiceaudio.c
> @@ -193,11 +193,6 @@ static int line_out_run (HWVoiceOut *hw, int live)
> return decr;
> }
>
> -static int line_out_write (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
> {
> SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
> @@ -326,11 +321,6 @@ static int line_in_run (HWVoiceIn *hw)
> return num_samples;
> }
>
> -static int line_in_read (SWVoiceIn *sw, void *buf, int size)
> -{
> - return audio_pcm_sw_read (sw, buf, size);
> -}
> -
> static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
> {
> SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
> @@ -378,13 +368,11 @@ static struct audio_pcm_ops audio_callbacks = {
> .init_out = line_out_init,
> .fini_out = line_out_fini,
> .run_out = line_out_run,
> - .write = line_out_write,
> .ctl_out = line_out_ctl,
>
> .init_in = line_in_init,
> .fini_in = line_in_fini,
> .run_in = line_in_run,
> - .read = line_in_read,
> .ctl_in = line_in_ctl,
> };
>
> diff --git a/audio/wavaudio.c b/audio/wavaudio.c
> index bbf3f3b346..17ab921cef 100644
> --- a/audio/wavaudio.c
> +++ b/audio/wavaudio.c
> @@ -84,11 +84,6 @@ static int wav_run_out (HWVoiceOut *hw, int live)
> return decr;
> }
>
> -static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
> -{
> - return audio_pcm_sw_write (sw, buf, len);
> -}
> -
> /* VICE code: Store number as little endian. */
> static void le_store (uint8_t *buf, uint32_t val, int len)
> {
> @@ -240,7 +235,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
> .init_out = wav_init_out,
> .fini_out = wav_fini_out,
> .run_out = wav_run_out,
> - .write = wav_write_out,
> .ctl_out = wav_ctl_out,
> };
>
> --
> 2.22.0
>
>
--
Marc-André Lureau
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops
2019-07-10 19:57 ` Marc-André Lureau
@ 2019-07-11 19:20 ` Zoltán Kővágó
0 siblings, 0 replies; 32+ messages in thread
From: Zoltán Kővágó @ 2019-07-11 19:20 UTC (permalink / raw)
To: Marc-André Lureau; +Cc: QEMU, Gerd Hoffmann
On 2019-07-10 21:57, Marc-André Lureau wrote:
> On Tue, Jul 9, 2019 at 10:57 PM Kővágó, Zoltán <dirty.ice.hu@gmail.com> wrote:
>>
>> They just called audio_pcm_sw_read/write anyway, so it makes no sense
>> to have them too. (The noaudio's read is the only exception, but it
>> should work with the generic code too.)
>
> It works with the generic code, but wouldn't it be more expensive?
It's a bit more expensive, but only if the guest has a sound card and
the guest is playing some sound, because otherwise the output is suspended.
> Perhaps there can be something in audio_pcm_sw_write() to skip the
> work if noaudio is the backend?
It feels like a hacky solution to the same problem what the function
pointer currently solves. On the other hand, it's unlikely any other
backend would drop every sample without processing them.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> audio/audio_int.h | 5 -----
>> audio/alsaaudio.c | 12 ------------
>> audio/audio.c | 8 ++++----
>> audio/coreaudio.c | 6 ------
>> audio/dsoundaudio.c | 12 ------------
>> audio/noaudio.c | 19 -------------------
>> audio/ossaudio.c | 12 ------------
>> audio/paaudio.c | 12 ------------
>> audio/sdlaudio.c | 6 ------
>> audio/spiceaudio.c | 12 ------------
>> audio/wavaudio.c | 6 ------
>> 11 files changed, 4 insertions(+), 106 deletions(-)
>>
>> diff --git a/audio/audio_int.h b/audio/audio_int.h
>> index 7e00c1332e..003b7ab8cc 100644
>> --- a/audio/audio_int.h
>> +++ b/audio/audio_int.h
>> @@ -150,13 +150,11 @@ struct audio_pcm_ops {
>> int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
>> void (*fini_out)(HWVoiceOut *hw);
>> int (*run_out) (HWVoiceOut *hw, int live);
>> - int (*write) (SWVoiceOut *sw, void *buf, int size);
>> int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
>>
>> int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
>> void (*fini_in) (HWVoiceIn *hw);
>> int (*run_in) (HWVoiceIn *hw);
>> - int (*read) (SWVoiceIn *sw, void *buf, int size);
>> int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
>> };
>>
>> @@ -210,11 +208,8 @@ 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_sw_write (SWVoiceOut *sw, void *buf, int len);
>> int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
>>
>> -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
>> -
>> int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
>> int live, int pending);
>>
>> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
>> index 3daa7c8f8f..e9e3a4819c 100644
>> --- a/audio/alsaaudio.c
>> +++ b/audio/alsaaudio.c
>> @@ -270,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw)
>> return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
>> }
>>
>> -static int alsa_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
>> {
>> switch (fmt) {
>> @@ -988,11 +983,6 @@ static int alsa_run_in (HWVoiceIn *hw)
>> return read_samples;
>> }
>>
>> -static int alsa_read (SWVoiceIn *sw, void *buf, int size)
>> -{
>> - return audio_pcm_sw_read (sw, buf, size);
>> -}
>> -
>> static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> {
>> ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
>> @@ -1076,13 +1066,11 @@ static struct audio_pcm_ops alsa_pcm_ops = {
>> .init_out = alsa_init_out,
>> .fini_out = alsa_fini_out,
>> .run_out = alsa_run_out,
>> - .write = alsa_write,
>> .ctl_out = alsa_ctl_out,
>>
>> .init_in = alsa_init_in,
>> .fini_in = alsa_fini_in,
>> .run_in = alsa_run_in,
>> - .read = alsa_read,
>> .ctl_in = alsa_ctl_in,
>> };
>>
>> diff --git a/audio/audio.c b/audio/audio.c
>> index d73cc086b6..b79f56fe64 100644
>> --- a/audio/audio.c
>> +++ b/audio/audio.c
>> @@ -594,7 +594,7 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
>> }
>> }
>>
>> -int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
>> +static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
>> {
>> HWVoiceIn *hw = sw->hw;
>> int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
>> @@ -696,7 +696,7 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
>> /*
>> * Soft voice (playback)
>> */
>> -int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
>> +static int audio_pcm_sw_write(SWVoiceOut *sw, void *buf, int size)
>> {
>> int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
>> int ret = 0, pos = 0, total = 0;
>> @@ -854,7 +854,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
>> return 0;
>> }
>>
>> - return sw->hw->pcm_ops->write(sw, buf, size);
>> + return audio_pcm_sw_write(sw, buf, size);
>> }
>>
>> int AUD_read (SWVoiceIn *sw, void *buf, int size)
>> @@ -869,7 +869,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size)
>> return 0;
>> }
>>
>> - return sw->hw->pcm_ops->read(sw, buf, size);
>> + return audio_pcm_sw_read(sw, buf, size);
>> }
>>
>> int AUD_get_buffer_size_out (SWVoiceOut *sw)
>> diff --git a/audio/coreaudio.c b/audio/coreaudio.c
>> index f0ab4014a8..091fe84a34 100644
>> --- a/audio/coreaudio.c
>> +++ b/audio/coreaudio.c
>> @@ -489,11 +489,6 @@ static OSStatus audioDeviceIOProc(
>> return 0;
>> }
>>
>> -static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
>> void *drv_opaque)
>> {
>> @@ -692,7 +687,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
>> .init_out = coreaudio_init_out,
>> .fini_out = coreaudio_fini_out,
>> .run_out = coreaudio_run_out,
>> - .write = coreaudio_write,
>> .ctl_out = coreaudio_ctl_out
>> };
>>
>> diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
>> index 07260f881e..11594c3095 100644
>> --- a/audio/dsoundaudio.c
>> +++ b/audio/dsoundaudio.c
>> @@ -454,11 +454,6 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
>> return 0;
>> }
>>
>> -static int dsound_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static int dsound_run_out (HWVoiceOut *hw, int live)
>> {
>> int err;
>> @@ -645,11 +640,6 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> return 0;
>> }
>>
>> -static int dsound_read (SWVoiceIn *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_read (sw, buf, len);
>> -}
>> -
>> static int dsound_run_in (HWVoiceIn *hw)
>> {
>> int err;
>> @@ -856,13 +846,11 @@ static struct audio_pcm_ops dsound_pcm_ops = {
>> .init_out = dsound_init_out,
>> .fini_out = dsound_fini_out,
>> .run_out = dsound_run_out,
>> - .write = dsound_write,
>> .ctl_out = dsound_ctl_out,
>>
>> .init_in = dsound_init_in,
>> .fini_in = dsound_fini_in,
>> .run_in = dsound_run_in,
>> - .read = dsound_read,
>> .ctl_in = dsound_ctl_in
>> };
>>
>> diff --git a/audio/noaudio.c b/audio/noaudio.c
>> index 14a0e4ab29..cbb02d9e49 100644
>> --- a/audio/noaudio.c
>> +++ b/audio/noaudio.c
>> @@ -61,11 +61,6 @@ static int no_run_out (HWVoiceOut *hw, int live)
>> return decr;
>> }
>>
>> -static int no_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write(sw, buf, len);
>> -}
>> -
>> static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
>> {
>> audio_pcm_init_info (&hw->info, as);
>> @@ -118,18 +113,6 @@ static int no_run_in (HWVoiceIn *hw)
>> return samples;
>> }
>>
>> -static int no_read (SWVoiceIn *sw, void *buf, int size)
>> -{
>> - /* use custom code here instead of audio_pcm_sw_read() to avoid
>> - * useless resampling/mixing */
>> - int samples = size >> sw->info.shift;
>> - int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
>> - 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;
>> -}
>> -
>> static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> {
>> (void) hw;
>> @@ -151,13 +134,11 @@ static struct audio_pcm_ops no_pcm_ops = {
>> .init_out = no_init_out,
>> .fini_out = no_fini_out,
>> .run_out = no_run_out,
>> - .write = no_write,
>> .ctl_out = no_ctl_out,
>>
>> .init_in = no_init_in,
>> .fini_in = no_fini_in,
>> .run_in = no_run_in,
>> - .read = no_read,
>> .ctl_in = no_ctl_in
>> };
>>
>> diff --git a/audio/ossaudio.c b/audio/ossaudio.c
>> index 456fba7480..b99edbec17 100644
>> --- a/audio/ossaudio.c
>> +++ b/audio/ossaudio.c
>> @@ -134,11 +134,6 @@ static void oss_poll_in (HWVoiceIn *hw)
>> qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
>> }
>>
>> -static int oss_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static int aud_to_ossfmt (AudioFormat fmt, int endianness)
>> {
>> switch (fmt) {
>> @@ -788,11 +783,6 @@ static int oss_run_in (HWVoiceIn *hw)
>> return read_samples;
>> }
>>
>> -static int oss_read (SWVoiceIn *sw, void *buf, int size)
>> -{
>> - return audio_pcm_sw_read (sw, buf, size);
>> -}
>> -
>> static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
>> {
>> OSSVoiceIn *oss = (OSSVoiceIn *) hw;
>> @@ -855,13 +845,11 @@ static struct audio_pcm_ops oss_pcm_ops = {
>> .init_out = oss_init_out,
>> .fini_out = oss_fini_out,
>> .run_out = oss_run_out,
>> - .write = oss_write,
>> .ctl_out = oss_ctl_out,
>>
>> .init_in = oss_init_in,
>> .fini_in = oss_fini_in,
>> .run_in = oss_run_in,
>> - .read = oss_read,
>> .ctl_in = oss_ctl_in
>> };
>>
>> diff --git a/audio/paaudio.c b/audio/paaudio.c
>> index d320d2e453..12471154ed 100644
>> --- a/audio/paaudio.c
>> +++ b/audio/paaudio.c
>> @@ -295,11 +295,6 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
>> return decr;
>> }
>>
>> -static int qpa_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> /* capture */
>> static void *qpa_thread_in (void *arg)
>> {
>> @@ -388,11 +383,6 @@ static int qpa_run_in (HWVoiceIn *hw)
>> return incr;
>> }
>>
>> -static int qpa_read (SWVoiceIn *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_read (sw, buf, len);
>> -}
>> -
>> static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
>> {
>> int format;
>> @@ -1016,13 +1006,11 @@ static struct audio_pcm_ops qpa_pcm_ops = {
>> .init_out = qpa_init_out,
>> .fini_out = qpa_fini_out,
>> .run_out = qpa_run_out,
>> - .write = qpa_write,
>> .ctl_out = qpa_ctl_out,
>>
>> .init_in = qpa_init_in,
>> .fini_in = qpa_fini_in,
>> .run_in = qpa_run_in,
>> - .read = qpa_read,
>> .ctl_in = qpa_ctl_in
>> };
>>
>> diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
>> index 42f7614124..c7fd487e0e 100644
>> --- a/audio/sdlaudio.c
>> +++ b/audio/sdlaudio.c
>> @@ -217,11 +217,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
>> }
>> }
>>
>> -static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static int sdl_run_out (HWVoiceOut *hw, int live)
>> {
>> int decr;
>> @@ -342,7 +337,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
>> .init_out = sdl_init_out,
>> .fini_out = sdl_fini_out,
>> .run_out = sdl_run_out,
>> - .write = sdl_write_out,
>> .ctl_out = sdl_ctl_out,
>> };
>>
>> diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
>> index 6f4a0558f8..6e7f13cffa 100644
>> --- a/audio/spiceaudio.c
>> +++ b/audio/spiceaudio.c
>> @@ -193,11 +193,6 @@ static int line_out_run (HWVoiceOut *hw, int live)
>> return decr;
>> }
>>
>> -static int line_out_write (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
>> {
>> SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
>> @@ -326,11 +321,6 @@ static int line_in_run (HWVoiceIn *hw)
>> return num_samples;
>> }
>>
>> -static int line_in_read (SWVoiceIn *sw, void *buf, int size)
>> -{
>> - return audio_pcm_sw_read (sw, buf, size);
>> -}
>> -
>> static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
>> {
>> SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
>> @@ -378,13 +368,11 @@ static struct audio_pcm_ops audio_callbacks = {
>> .init_out = line_out_init,
>> .fini_out = line_out_fini,
>> .run_out = line_out_run,
>> - .write = line_out_write,
>> .ctl_out = line_out_ctl,
>>
>> .init_in = line_in_init,
>> .fini_in = line_in_fini,
>> .run_in = line_in_run,
>> - .read = line_in_read,
>> .ctl_in = line_in_ctl,
>> };
>>
>> diff --git a/audio/wavaudio.c b/audio/wavaudio.c
>> index bbf3f3b346..17ab921cef 100644
>> --- a/audio/wavaudio.c
>> +++ b/audio/wavaudio.c
>> @@ -84,11 +84,6 @@ static int wav_run_out (HWVoiceOut *hw, int live)
>> return decr;
>> }
>>
>> -static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
>> -{
>> - return audio_pcm_sw_write (sw, buf, len);
>> -}
>> -
>> /* VICE code: Store number as little endian. */
>> static void le_store (uint8_t *buf, uint32_t val, int len)
>> {
>> @@ -240,7 +235,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
>> .init_out = wav_init_out,
>> .fini_out = wav_fini_out,
>> .run_out = wav_run_out,
>> - .write = wav_write_out,
>> .ctl_out = wav_ctl_out,
>> };
>>
>> --
>> 2.22.0
>>
>>
>
>
^ permalink raw reply [flat|nested] 32+ messages in thread
* [Qemu-devel] [PATCH 11/11] audio: use size_t where makes sense
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (9 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 10/11] audio: remove read and write pcm_ops Kővágó, Zoltán
@ 2019-07-09 18:35 ` Kővágó, Zoltán
2019-07-09 22:04 ` [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends no-reply
2019-07-09 22:44 ` no-reply
12 siblings, 0 replies; 32+ messages in thread
From: Kővágó, Zoltán @ 2019-07-09 18:35 UTC (permalink / raw)
To: qemu-devel; +Cc: Paolo Bonzini, Gerd Hoffmann, Pavel Dovgalyuk
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.h | 4 +-
audio/audio_int.h | 26 +++----
audio/audio_template.h | 14 ++--
audio/mixeng.h | 9 +--
audio/rate_template.h | 2 +-
include/sysemu/replay.h | 4 +-
audio/alsaaudio.c | 26 +++----
audio/audio.c | 156 ++++++++++++++++++++--------------------
audio/coreaudio.c | 10 +--
audio/dsoundaudio.c | 11 +--
audio/noaudio.c | 16 ++---
audio/ossaudio.c | 45 ++++++------
audio/paaudio.c | 44 ++++++------
audio/sdlaudio.c | 20 +++---
audio/spiceaudio.c | 12 ++--
audio/wavaudio.c | 8 +--
replay/replay-audio.c | 16 ++---
replay/replay.c | 2 +-
18 files changed, 212 insertions(+), 213 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index 4a95758516..2db27bba7b 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -112,7 +112,7 @@ SWVoiceOut *AUD_open_out (
);
void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
-int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+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);
@@ -133,7 +133,7 @@ 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 003b7ab8cc..a674c5374a 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -61,12 +61,12 @@ 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;
@@ -82,13 +82,13 @@ 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;
@@ -103,7 +103,7 @@ 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;
@@ -120,7 +120,7 @@ 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;
@@ -149,12 +149,12 @@ struct audio_driver {
struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
- int (*run_out) (HWVoiceOut *hw, int live);
+ size_t (*run_out)(HWVoiceOut *hw, size_t live);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
- int (*run_in) (HWVoiceIn *hw);
+ size_t (*run_in)(HWVoiceIn *hw);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
@@ -208,10 +208,10 @@ 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);
@@ -224,7 +224,7 @@ 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 54f07338e7..2562bf5f00 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -75,16 +75,16 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
HWBUF = NULL;
}
-static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
+static bool glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
if (!HWBUF) {
- dolog ("Could not allocate " NAME " buffer (%d samples)\n",
- hw->samples);
- return -1;
+ dolog("Could not allocate " NAME " buffer (%zu samples)\n",
+ hw->samples);
+ return false;
}
- return 0;
+ return true;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -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;
}
@@ -279,7 +279,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
- if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
+ if (!glue(audio_pcm_hw_alloc_resources_, TYPE)(hw)) {
goto err1;
}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index b53a5ef99a..18e62c7c49 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -33,6 +33,7 @@ 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 +42,10 @@ 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/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/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/audio/alsaaudio.c b/audio/alsaaudio.c
index e9e3a4819c..591344dccd 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -681,10 +681,10 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
}
}
-static int alsa_run_out (HWVoiceOut *hw, int live)
+static size_t alsa_run_out(HWVoiceOut *hw, size_t live)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- int decr;
+ size_t decr;
snd_pcm_sframes_t avail;
avail = alsa_get_avail (alsa->handle);
@@ -739,8 +739,8 @@ 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;
}
@@ -841,8 +841,8 @@ 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;
}
@@ -863,17 +863,17 @@ static void alsa_fini_in (HWVoiceIn *hw)
alsa->pcm_buf = NULL;
}
-static int alsa_run_in (HWVoiceIn *hw)
+static size_t alsa_run_in(HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
int hwshift = hw->info.shift;
int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int decr;
+ size_t live = audio_pcm_hw_get_live_in (hw);
+ size_t dead = hw->samples - live;
+ size_t decr;
struct {
- int add;
- int len;
+ size_t add;
+ size_t len;
} bufs[2] = {
{ .add = hw->wpos, .len = 0 },
{ .add = 0, .len = 0 }
@@ -913,7 +913,7 @@ static int alsa_run_in (HWVoiceIn *hw)
}
}
- decr = MIN (dead, avail);
+ decr = MIN(dead, avail);
if (!decr) {
return 0;
}
diff --git a/audio/audio.c b/audio/audio.c
index b79f56fe64..a497cb024e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -526,10 +526,10 @@ 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) {
@@ -539,28 +539,28 @@ 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);
@@ -574,14 +574,14 @@ 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;
}
@@ -594,17 +594,17 @@ 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;
}
@@ -618,9 +618,9 @@ 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;
}
@@ -629,11 +629,6 @@ 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;
@@ -654,10 +649,10 @@ 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) {
@@ -671,9 +666,9 @@ 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);
@@ -682,10 +677,10 @@ 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;
@@ -696,10 +691,10 @@ 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;
@@ -708,8 +703,8 @@ 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;
}
@@ -763,7 +758,7 @@ 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,
@@ -842,7 +837,7 @@ 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 */
@@ -857,7 +852,7 @@ 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 */
@@ -966,17 +961,17 @@ 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;
}
@@ -989,9 +984,9 @@ 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;
@@ -999,8 +994,8 @@ 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;
}
@@ -1015,9 +1010,10 @@ 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;
@@ -1028,17 +1024,17 @@ 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;
@@ -1047,9 +1043,9 @@ 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)
@@ -1058,16 +1054,16 @@ 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;
}
@@ -1102,13 +1098,13 @@ 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) {
@@ -1123,8 +1119,8 @@ 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;
}
@@ -1164,7 +1160,7 @@ 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);
@@ -1179,7 +1175,7 @@ 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) {
@@ -1195,15 +1191,15 @@ 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;
@@ -1226,8 +1222,8 @@ 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;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 091fe84a34..d1be58b40a 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,9 +43,9 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
- int live;
- int decr;
- int rpos;
+ size_t live;
+ size_t decr;
+ size_t rpos;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -397,9 +397,9 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
return 0;
}
-static int coreaudio_run_out (HWVoiceOut *hw, int live)
+static size_t coreaudio_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
if (coreaudio_lock (core, "coreaudio_run_out")) {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 11594c3095..67aa1d0ad5 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -454,19 +454,20 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int dsound_run_out (HWVoiceOut *hw, int live)
+static size_t dsound_run_out(HWVoiceOut *hw, size_t live)
{
int err;
HRESULT hr;
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
- int len, hwshift;
+ size_t len;
+ int hwshift;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
DWORD wpos, ppos, old_pos;
LPVOID p1, p2;
- int bufsize;
+ size_t bufsize;
dsound *s = ds->s;
AudiodevDsoundOptions *dso = &s->dev->u.dsound;
@@ -640,13 +641,13 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static int dsound_run_in (HWVoiceIn *hw)
+static size_t dsound_run_in(HWVoiceIn *hw)
{
int err;
HRESULT hr;
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
- int live, len, dead;
+ size_t live, len, dead;
DWORD blen1, blen2;
DWORD len1, len2;
DWORD decr;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index cbb02d9e49..0fb2629cf2 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -41,10 +41,10 @@ typedef struct NoVoiceIn {
int64_t old_ticks;
} NoVoiceIn;
-static int no_run_out (HWVoiceOut *hw, int live)
+static size_t no_run_out(HWVoiceOut *hw, size_t live)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- int decr, samples;
+ size_t decr, samples;
int64_t now;
int64_t ticks;
int64_t bytes;
@@ -52,7 +52,7 @@ 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 = MIN(bytes, INT_MAX);
+ bytes = MIN(bytes, SIZE_MAX);
samples = bytes >> hw->info.shift;
no->old_ticks = now;
@@ -92,12 +92,12 @@ static void no_fini_in (HWVoiceIn *hw)
(void) hw;
}
-static int no_run_in (HWVoiceIn *hw)
+static size_t no_run_in(HWVoiceIn *hw)
{
NoVoiceIn *no = (NoVoiceIn *) hw;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int samples = 0;
+ size_t live = audio_pcm_hw_get_live_in(hw);
+ size_t dead = hw->samples - live;
+ size_t samples = 0;
if (dead) {
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
@@ -106,7 +106,7 @@ static int no_run_in (HWVoiceIn *hw)
muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
no->old_ticks = now;
- bytes = MIN (bytes, INT_MAX);
+ bytes = MIN (bytes, SIZE_MAX);
samples = bytes >> hw->info.shift;
samples = MIN (samples, dead);
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index b99edbec17..1696933688 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -411,13 +411,14 @@ static void oss_write_pending (OSSVoiceOut *oss)
}
}
-static int oss_run_out (HWVoiceOut *hw, int live)
+static size_t oss_run_out(HWVoiceOut *hw, size_t live)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- int err, decr;
+ int err;
+ size_t decr;
struct audio_buf_info abinfo;
struct count_info cntinfo;
- int bufsize;
+ size_t bufsize;
bufsize = hw->samples << hw->info.shift;
@@ -476,8 +477,8 @@ 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 +544,8 @@ 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 +569,8 @@ 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 +582,7 @@ 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 +694,8 @@ 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;
}
@@ -714,17 +715,17 @@ static void oss_fini_in (HWVoiceIn *hw)
oss->pcm_buf = NULL;
}
-static int oss_run_in (HWVoiceIn *hw)
+static size_t oss_run_in(HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
int hwshift = hw->info.shift;
int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
+ size_t live = audio_pcm_hw_get_live_in (hw);
+ size_t dead = hw->samples - live;
size_t read_samples = 0;
struct {
- int add;
- int len;
+ size_t add;
+ size_t len;
} bufs[2] = {
{ .add = hw->wpos, .len = 0 },
{ .add = 0, .len = 0 }
@@ -751,9 +752,9 @@ static int oss_run_in (HWVoiceIn *hw)
if (nread > 0) {
if (nread & hw->info.align) {
- dolog ("warning: Misaligned read %zd (requested %d), "
- "alignment %d\n", nread, bufs[i].add << hwshift,
- hw->info.align + 1);
+ dolog("warning: Misaligned read %zd (requested %zu), "
+ "alignment %d\n", nread, bufs[i].add << hwshift,
+ hw->info.align + 1);
}
read_samples += nread >> hwshift;
hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift);
@@ -766,9 +767,9 @@ static int oss_run_in (HWVoiceIn *hw)
case EAGAIN:
break;
default:
- oss_logerr (
+ oss_logerr(
errno,
- "Failed to read %d bytes of audio (to %p)\n",
+ "Failed to read %zu bytes of audio (to %p)\n",
bufs[i].len, p
);
break;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 12471154ed..dfe89d9ee5 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -30,30 +30,30 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
- int done;
- int live;
- int decr;
- int rpos;
+ size_t done;
+ size_t live;
+ size_t decr;
+ size_t rpos;
pa_stream *stream;
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
- int samples;
+ size_t samples;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
- int done;
- int dead;
- int incr;
- int wpos;
+ size_t done;
+ size_t dead;
+ size_t incr;
+ size_t wpos;
pa_stream *stream;
void *pcm_buf;
struct audio_pt pt;
const void *read_data;
size_t read_index, read_length;
paaudio *g;
- int samples;
+ size_t samples;
} PAVoiceIn;
static void qpa_conn_fini(PAConnection *c);
@@ -219,7 +219,7 @@ static void *qpa_thread_out (void *arg)
}
for (;;) {
- int decr, to_mix, rpos;
+ size_t decr, to_mix, rpos;
for (;;) {
if (pa->done) {
@@ -244,7 +244,7 @@ static void *qpa_thread_out (void *arg)
while (to_mix) {
int error;
- int chunk = MIN (to_mix, hw->samples - rpos);
+ size_t chunk = MIN (to_mix, hw->samples - rpos);
struct st_sample *src = hw->mix_buf + rpos;
hw->clip (pa->pcm_buf, src, chunk);
@@ -273,9 +273,9 @@ static void *qpa_thread_out (void *arg)
return NULL;
}
-static int qpa_run_out (HWVoiceOut *hw, int live)
+static size_t qpa_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
PAVoiceOut *pa = (PAVoiceOut *) hw;
if (audio_pt_lock(&pa->pt, __func__)) {
@@ -306,7 +306,7 @@ static void *qpa_thread_in (void *arg)
}
for (;;) {
- int incr, to_grab, wpos;
+ size_t incr, to_grab, wpos;
for (;;) {
if (pa->done) {
@@ -331,7 +331,7 @@ static void *qpa_thread_in (void *arg)
while (to_grab) {
int error;
- int chunk = MIN (to_grab, hw->samples - wpos);
+ size_t chunk = MIN (to_grab, hw->samples - wpos);
void *buf = advance (pa->pcm_buf, wpos);
if (qpa_simple_read (pa, buf,
@@ -359,9 +359,9 @@ static void *qpa_thread_in (void *arg)
return NULL;
}
-static int qpa_run_in (HWVoiceIn *hw)
+static size_t qpa_run_in(HWVoiceIn *hw)
{
- int live, incr, dead;
+ size_t live, incr, dead;
PAVoiceIn *pa = (PAVoiceIn *) hw;
if (audio_pt_lock(&pa->pt, __func__)) {
@@ -582,8 +582,8 @@ 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;
}
@@ -650,8 +650,8 @@ 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/sdlaudio.c b/audio/sdlaudio.c
index c7fd487e0e..14b11f0335 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -41,8 +41,8 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
- int live;
- int decr;
+ size_t live;
+ size_t decr;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -184,22 +184,22 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
- int samples = len >> hw->info.shift;
- int to_mix, decr;
+ size_t samples = len >> hw->info.shift;
+ size_t to_mix, decr;
if (s->exit || !sdl->live) {
return;
}
- /* dolog ("in callback samples=%d live=%d\n", samples, sdl->live); */
+ /* dolog ("in callback samples=%zu live=%zu\n", samples, sdl->live); */
to_mix = MIN(samples, sdl->live);
decr = to_mix;
while (to_mix) {
- int chunk = MIN(to_mix, hw->samples - hw->rpos);
+ size_t 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); */
+ /* dolog ("in callback to_mix %zu, chunk %zu\n", to_mix, chunk); */
hw->clip(buf, src, chunk);
hw->rpos = (hw->rpos + chunk) % hw->samples;
to_mix -= chunk;
@@ -209,7 +209,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
sdl->live -= decr;
sdl->decr += decr;
- /* dolog ("done len=%d\n", len); */
+ /* dolog ("done len=%zu\n", len); */
/* SDL2 does not clear the remaining buffer for us, so do it on our own */
if (samples) {
@@ -217,9 +217,9 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
}
-static int sdl_run_out (HWVoiceOut *hw, int live)
+static size_t sdl_run_out(HWVoiceOut *hw, size_t live)
{
- int decr;
+ size_t decr;
SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDL_LockAudio();
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6e7f13cffa..d9db1a39e6 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -153,11 +153,11 @@ static void line_out_fini (HWVoiceOut *hw)
spice_server_remove_interface (&out->sin.base);
}
-static int line_out_run (HWVoiceOut *hw, int live)
+static size_t line_out_run (HWVoiceOut *hw, size_t live)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- int rpos, decr;
- int samples;
+ size_t rpos, decr;
+ size_t samples;
if (!live) {
return 0;
@@ -276,12 +276,12 @@ static void line_in_fini (HWVoiceIn *hw)
spice_server_remove_interface (&in->sin.base);
}
-static int line_in_run (HWVoiceIn *hw)
+static size_t line_in_run(HWVoiceIn *hw)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- int num_samples;
+ size_t num_samples;
int ready;
- int len[2];
+ size_t len[2];
uint64_t delta_samp;
const uint32_t *samples;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 17ab921cef..b6eeeb4e26 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -40,10 +40,10 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-static int wav_run_out (HWVoiceOut *hw, int live)
+static size_t wav_run_out(HWVoiceOut *hw, size_t live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int rpos, decr, samples;
+ size_t rpos, decr, samples;
uint8_t *dst;
struct st_sample *src;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
@@ -139,8 +139,8 @@ 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/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();
diff --git a/replay/replay.c b/replay/replay.c
index 8b172b2d1b..a1135a9943 100644
--- a/replay/replay.c
+++ b/replay/replay.c
@@ -22,7 +22,7 @@
/* Current version of the replay mechanism.
Increase it when file format changes. */
-#define REPLAY_VERSION 0xe02007
+#define REPLAY_VERSION 0xe02008
/* Size of replay log header */
#define HEADER_SIZE (sizeof(uint32_t) + sizeof(uint64_t))
--
2.22.0
^ permalink raw reply related [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (10 preceding siblings ...)
2019-07-09 18:35 ` [Qemu-devel] [PATCH 11/11] audio: use size_t where makes sense Kővágó, Zoltán
@ 2019-07-09 22:04 ` no-reply
2019-07-09 22:44 ` no-reply
12 siblings, 0 replies; 32+ messages in thread
From: no-reply @ 2019-07-09 22:04 UTC (permalink / raw)
To: dirty.ice.hu; +Cc: qemu-devel, kraxel
Patchew URL: https://patchew.org/QEMU/cover.1562695780.git.DirtY.iCE.hu@gmail.com/
Hi,
This series seems to have some coding style problems. See output below for
more information:
Type: series
Message-id: cover.1562695780.git.DirtY.iCE.hu@gmail.com
Subject: [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends
=== TEST SCRIPT BEGIN ===
#!/bin/bash
git rev-parse base > /dev/null || exit 0
git config --local diff.renamelimit 0
git config --local diff.renames True
git config --local diff.algorithm histogram
./scripts/checkpatch.pl --mailback base..
=== TEST SCRIPT END ===
Updating 3c8cf5a9c21ff8782164d1def7f44bd888713384
From https://github.com/patchew-project/qemu
* [new tag] patchew/20190709213651.77315-1-mrolnik@gmail.com -> patchew/20190709213651.77315-1-mrolnik@gmail.com
Switched to a new branch 'test'
3080c9b audio: use size_t where makes sense
c4faae9 audio: remove read and write pcm_ops
8d1cbe2 paaudio: fix playback glitches
dbeb913 audio: do not run each backend in audio_run
ec4e34e audio: remove audio_MIN, audio_MAX
b36085a paaudio: properly disconnect streams in fini_*
db53be0 paaudio: do not move stream when sink/source name is specified
052ab5c audio: audiodev= parameters no longer optional when -audiodev present
c0cf5a5 audio: add audiodev properties to frontends
15c571f audio: basic support for multi backend audio
88dfd06 audio: reduce glob_audio_state usage
=== OUTPUT BEGIN ===
1/11 Checking commit 88dfd065d232 (audio: reduce glob_audio_state usage)
2/11 Checking commit 15c571f4acb7 (audio: basic support for multi backend audio)
ERROR: space prohibited between function name and open parenthesis '('
#168: FILE: audio/audio.c:1526:
+ QLIST_INSERT_HEAD (&card->state->card_head, card, entries);
total: 1 errors, 0 warnings, 366 lines checked
Patch 2/11 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
3/11 Checking commit c0cf5a5363ac (audio: add audiodev properties to frontends)
4/11 Checking commit 052ab5cd93bd (audio: audiodev= parameters no longer optional when -audiodev present)
5/11 Checking commit db53be05092d (paaudio: do not move stream when sink/source name is specified)
6/11 Checking commit b36085a0bf72 (paaudio: properly disconnect streams in fini_*)
7/11 Checking commit ec4e34eade3d (audio: remove audio_MIN, audio_MAX)
ERROR: space prohibited between function name and open parenthesis '('
#22: FILE: audio/alsaaudio.c:637:
+ int len = MIN (alsa->pending, left_till_end_samples);
ERROR: space prohibited between function name and open parenthesis '('
#31: FILE: audio/alsaaudio.c:700:
+ decr = MIN (live, avail);
ERROR: space prohibited between function name and open parenthesis '('
#40: FILE: audio/alsaaudio.c:918:
+ decr = MIN (dead, avail);
ERROR: space prohibited between function name and open parenthesis '('
#53: FILE: audio/audio.c:536:
+ m = MIN (m, sw->total_hw_samples_acquired);
ERROR: space prohibited between function name and open parenthesis '('
#62: FILE: audio/audio.c:556:
+ int len = MIN (left, live);
ERROR: space prohibited between function name and open parenthesis '('
#70: FILE: audio/audio.c:563:
+ int samples_to_clip = MIN (len, samples_till_end_of_buf);
ERROR: space prohibited between function name and open parenthesis '('
#79: FILE: audio/audio.c:617:
+ swlim = MIN (swlim, samples);
ERROR: space prohibited between function name and open parenthesis '('
#88: FILE: audio/audio.c:665:
+ m = MIN (m, sw->total_hw_samples_mixed);
ERROR: space prohibited between function name and open parenthesis '('
#97: FILE: audio/audio.c:728:
+ swlim = MIN (swlim, samples);
ERROR: space prohibited between function name and open parenthesis '('
#106: FILE: audio/audio.c:740:
+ blck = MIN (dead, left);
ERROR: space prohibited between function name and open parenthesis '('
#115: FILE: audio/audio.c:1032:
+ int to_write = MIN (till_end_of_hw, n);
ERROR: space prohibited between function name and open parenthesis '('
#124: FILE: audio/audio.c:1050:
+ n = MIN (samples, hw->samples - rpos);
ERROR: space prohibited between function name and open parenthesis '('
#133: FILE: audio/audio.c:1206:
+ int to_capture = MIN (live, left);
ERROR: space prohibited between function name and open parenthesis '('
#174: FILE: audio/coreaudio.c:416:
+ decr = MIN (core->decr, live);
ERROR: space prohibited between function name and open parenthesis '('
#187: FILE: audio/dsoundaudio.c:710:
+ len = MIN (len, dead);
ERROR: space prohibited between function name and open parenthesis '('
#205: FILE: audio/noaudio.c:59:
+ decr = MIN (live, samples);
ERROR: space prohibited between function name and open parenthesis '('
#214: FILE: audio/noaudio.c:114:
+ bytes = MIN (bytes, INT_MAX);
ERROR: space prohibited between function name and open parenthesis '('
#217: FILE: audio/noaudio.c:116:
+ samples = MIN (samples, dead);
ERROR: space prohibited between function name and open parenthesis '('
#226: FILE: audio/noaudio.c:127:
+ int to_clear = MIN (samples, total);
ERROR: space prohibited between function name and open parenthesis '('
#239: FILE: audio/ossaudio.c:391:
+ int samples_to_write = MIN (oss->pending, samples_till_end);
ERROR: space prohibited between function name and open parenthesis '('
#248: FILE: audio/ossaudio.c:440:
+ decr = MIN (bytes >> hw->info.shift, live);
ERROR: space prohibited between function name and open parenthesis '('
#257: FILE: audio/ossaudio.c:459:
+ decr = MIN (abinfo.bytes >> hw->info.shift, live);
ERROR: space prohibited between function name and open parenthesis '('
#279: FILE: audio/paaudio.c:247:
+ int chunk = MIN (to_mix, hw->samples - rpos);
ERROR: space prohibited between function name and open parenthesis '('
#288: FILE: audio/paaudio.c:285:
+ decr = MIN (live, pa->decr);
ERROR: space prohibited between function name and open parenthesis '('
#306: FILE: audio/paaudio.c:339:
+ int chunk = MIN (to_grab, hw->samples - wpos);
ERROR: space prohibited between function name and open parenthesis '('
#315: FILE: audio/paaudio.c:378:
+ incr = MIN (dead, pa->incr);
ERROR: space prohibited between function name and open parenthesis '('
#341: FILE: audio/sdlaudio.c:239:
+ decr = MIN (sdl->decr, live);
ERROR: space prohibited between function name and open parenthesis '('
#354: FILE: audio/spiceaudio.c:167:
+ decr = MIN (live, decr);
ERROR: space prohibited between function name and open parenthesis '('
#361: FILE: audio/spiceaudio.c:173:
+ int len = MIN (samples, left_till_end_samples);
ERROR: space prohibited between function name and open parenthesis '('
#369: FILE: audio/spiceaudio.c:180:
+ len = MIN (len, out->fsize);
ERROR: space prohibited between function name and open parenthesis '('
#378: FILE: audio/spiceaudio.c:298:
+ num_samples = MIN (num_samples, delta_samp);
ERROR: space prohibited between function name and open parenthesis '('
#387: FILE: audio/spiceaudio.c:308:
+ num_samples = MIN (ready, num_samples);
ERROR: space prohibited between function name and open parenthesis '('
#400: FILE: audio/wavaudio.c:62:
+ decr = MIN (live, samples);
ERROR: space prohibited between function name and open parenthesis '('
#406: FILE: audio/wavaudio.c:67:
+ int convert_samples = MIN (samples, left_till_end_samples);
ERROR: space prohibited between function name and open parenthesis '('
#419: FILE: hw/audio/ac97.c:967:
+ temp = MIN (temp, max);
ERROR: space prohibited between function name and open parenthesis '('
#428: FILE: hw/audio/ac97.c:976:
+ to_copy = MIN (temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#428: FILE: hw/audio/ac97.c:976:
+ to_copy = MIN (temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#437: FILE: hw/audio/ac97.c:1022:
+ int temp = MIN (elapsed, sizeof (s->silence));
ERROR: space prohibited between function name and open parenthesis '('
#437: FILE: hw/audio/ac97.c:1022:
+ int temp = MIN (elapsed, sizeof (s->silence));
ERROR: space prohibited between function name and open parenthesis '('
#446: FILE: hw/audio/ac97.c:1043:
+ temp = MIN (temp, max);
ERROR: space prohibited between function name and open parenthesis '('
#455: FILE: hw/audio/ac97.c:1052:
+ to_copy = MIN (temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#455: FILE: hw/audio/ac97.c:1052:
+ to_copy = MIN (temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#468: FILE: hw/audio/adlib.c:198:
+ to_play = MIN (s->left, samples);
ERROR: space prohibited between function name and open parenthesis '('
#477: FILE: hw/audio/adlib.c:213:
+ samples = MIN (samples, s->samples - s->pos);
ERROR: space prohibited between function name and open parenthesis '('
#490: FILE: hw/audio/cs4231a.c:538:
+ to_copy = MIN (temp, left);
ERROR: space prohibited between function name and open parenthesis '('
#499: FILE: hw/audio/cs4231a.c:581:
+ copy = MIN (till, copy);
ERROR: space prohibited between function name and open parenthesis '('
#512: FILE: hw/audio/es1370.c:648:
+ int temp = MIN (max, MIN (left, csc_bytes));
ERROR: space prohibited between function name and open parenthesis '('
#512: FILE: hw/audio/es1370.c:648:
+ int temp = MIN (max, MIN (left, csc_bytes));
ERROR: space prohibited between function name and open parenthesis '('
#521: FILE: hw/audio/es1370.c:657:
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#521: FILE: hw/audio/es1370.c:657:
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#530: FILE: hw/audio/es1370.c:675:
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#530: FILE: hw/audio/es1370.c:675:
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#543: FILE: hw/audio/gus.c:120:
+ to_play = MIN (samples, s->left);
ERROR: space prohibited between function name and open parenthesis '('
#552: FILE: hw/audio/gus.c:135:
+ samples = MIN (samples, s->samples);
ERROR: space prohibited between function name and open parenthesis '('
#561: FILE: hw/audio/gus.c:195:
+ int to_copy = MIN ((size_t) left, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#561: FILE: hw/audio/gus.c:195:
+ int to_copy = MIN ((size_t) left, sizeof (tmpbuf));
ERROR: space prohibited between function name and open parenthesis '('
#691: FILE: hw/audio/sb16.c:1171:
+ to_copy = MIN (temp, left);
total: 57 errors, 0 warnings, 559 lines checked
Patch 7/11 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
8/11 Checking commit dbeb913b9bda (audio: do not run each backend in audio_run)
9/11 Checking commit 8d1cbe2aac33 (paaudio: fix playback glitches)
10/11 Checking commit c4faae967433 (audio: remove read and write pcm_ops)
11/11 Checking commit 3080c9b66a87 (audio: use size_t where makes sense)
ERROR: space prohibited between function name and open parenthesis '('
#62: FILE: audio/alsaaudio.c:871:
+ size_t live = audio_pcm_hw_get_live_in (hw);
ERROR: space prohibited between function name and open parenthesis '('
#91: FILE: audio/audio.c:529:
+static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
ERROR: space prohibited between function name and open parenthesis '('
#109: FILE: audio/audio.c:544:
+ size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
ERROR: space prohibited between function name and open parenthesis '('
#126: FILE: audio/audio.c:556:
+ size_t len = MIN (left, live);
ERROR: space prohibited between function name and open parenthesis '('
#135: FILE: audio/audio.c:563:
+ size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
ERROR: space prohibited between function name and open parenthesis '('
#210: FILE: audio/audio.c:652:
+static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
ERROR: space prohibited between function name and open parenthesis '('
#223: FILE: audio/audio.c:669:
+static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
WARNING: line over 80 characters
#253: FILE: audio/audio.c:696:
+ size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
ERROR: space prohibited between function name and open parenthesis '('
#301: FILE: audio/audio.c:964:
+static size_t audio_get_avail (SWVoiceIn *sw)
ERROR: space prohibited between function name and open parenthesis '('
#409: FILE: audio/audio.c:1066:
+ dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
ERROR: space prohibited between function name and open parenthesis '('
#795: FILE: audio/noaudio.c:109:
+ bytes = MIN (bytes, SIZE_MAX);
ERROR: space prohibited between function name and open parenthesis '('
#886: FILE: audio/ossaudio.c:723:
+ size_t live = audio_pcm_hw_get_live_in (hw);
ERROR: space prohibited between function name and open parenthesis '('
#981: FILE: audio/paaudio.c:247:
+ size_t chunk = MIN (to_mix, hw->samples - rpos);
ERROR: space prohibited between function name and open parenthesis '('
#1011: FILE: audio/paaudio.c:334:
+ size_t chunk = MIN (to_grab, hw->samples - wpos);
ERROR: space prohibited between function name and open parenthesis '('
#1135: FILE: audio/spiceaudio.c:156:
+static size_t line_out_run (HWVoiceOut *hw, size_t live)
WARNING: line over 80 characters
#1201: FILE: include/sysemu/replay.h:184:
+void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size);
total: 14 errors, 2 warnings, 1106 lines checked
Patch 11/11 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
=== OUTPUT END ===
Test command exited with code: 1
The full log is available at
http://patchew.org/logs/cover.1562695780.git.DirtY.iCE.hu@gmail.com/testing.checkpatch/?type=message.
---
Email generated automatically by Patchew [https://patchew.org/].
Please send your feedback to patchew-devel@redhat.com
^ permalink raw reply [flat|nested] 32+ messages in thread* Re: [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends
2019-07-09 18:35 [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends Kővágó, Zoltán
` (11 preceding siblings ...)
2019-07-09 22:04 ` [Qemu-devel] [PATCH 00/11] Multiple simultaneous audio backends no-reply
@ 2019-07-09 22:44 ` no-reply
12 siblings, 0 replies; 32+ messages in thread
From: no-reply @ 2019-07-09 22:44 UTC (permalink / raw)
To: dirty.ice.hu; +Cc: qemu-devel, kraxel
Patchew URL: https://patchew.org/QEMU/cover.1562695780.git.DirtY.iCE.hu@gmail.com/
Hi,
This series failed the asan build test. Please find the testing commands and
their output below. If you have Docker installed, you can probably reproduce it
locally.
=== TEST SCRIPT BEGIN ===
#!/bin/bash
make docker-image-fedora V=1 NETWORK=1
time make docker-test-debug@fedora TARGET_LIST=x86_64-softmmu J=14 NETWORK=1
=== TEST SCRIPT END ===
PASS 5 test-qmp-cmds /qmp/dispatch_cmd_success_response
PASS 6 test-qmp-cmds /qmp/dealloc_types
PASS 7 test-qmp-cmds /qmp/dealloc_partial
==9597==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-string-input-visitor -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-string-input-visitor"
PASS 4 fdc-test /x86_64/fdc/media_change
PASS 5 fdc-test /x86_64/fdc/sense_interrupt
---
PASS 32 test-opts-visitor /visitor/opts/range/beyond
PASS 33 test-opts-visitor /visitor/opts/dict/unvisited
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-coroutine -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-coroutine"
==9635==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==9635==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffc4c709000; bottom 0x7f5cdbaf8000; size: 0x009f70c11000 (684791500800)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 1 test-coroutine /basic/no-dangling-access
---
PASS 12 test-aio /aio/event/flush
PASS 13 test-aio /aio/event/wait/no-flush-cb
PASS 11 fdc-test /x86_64/fdc/read_no_dma_18
==9652==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 14 test-aio /aio/timer/schedule
PASS 15 test-aio /aio/coroutine/queue-chaining
PASS 16 test-aio /aio-gsource/flush
---
PASS 27 test-aio /aio-gsource/event/wait/no-flush-cb
PASS 28 test-aio /aio-gsource/timer/schedule
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-aio-multithread -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-aio-multithread"
==9658==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-aio-multithread /aio/multi/lifecycle
PASS 2 test-aio-multithread /aio/multi/schedule
PASS 12 fdc-test /x86_64/fdc/read_no_dma_19
PASS 13 fdc-test /x86_64/fdc/fuzz-registers
PASS 3 test-aio-multithread /aio/multi/mutex/contended
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/ide-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="ide-test"
==9686==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 ide-test /x86_64/ide/identify
==9692==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 ide-test /x86_64/ide/flush
==9698==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 ide-test /x86_64/ide/bmdma/simple_rw
PASS 4 test-aio-multithread /aio/multi/mutex/handoff
==9704==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 4 ide-test /x86_64/ide/bmdma/trim
PASS 5 test-aio-multithread /aio/multi/mutex/mcs
==9715==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 5 ide-test /x86_64/ide/bmdma/short_prdt
PASS 6 test-aio-multithread /aio/multi/mutex/pthread
==9726==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-throttle -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-throttle"
PASS 1 test-throttle /throttle/leak_bucket
PASS 2 test-throttle /throttle/compute_wait
---
PASS 6 test-throttle /throttle/detach_attach
PASS 7 test-throttle /throttle/config_functions
PASS 8 test-throttle /throttle/accounting
==9733==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 9 test-throttle /throttle/groups
PASS 10 test-throttle /throttle/config/enabled
PASS 11 test-throttle /throttle/config/conflicting
---
PASS 15 test-throttle /throttle/config/iops_size
PASS 6 ide-test /x86_64/ide/bmdma/one_sector_short_prdt
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-thread-pool -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-thread-pool"
==9740==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-thread-pool /thread-pool/submit
PASS 2 test-thread-pool /thread-pool/submit-aio
PASS 3 test-thread-pool /thread-pool/submit-co
PASS 4 test-thread-pool /thread-pool/submit-many
==9738==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 7 ide-test /x86_64/ide/bmdma/long_prdt
==9749==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==9749==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffff7d00000; bottom 0x7fe50fffe000; size: 0x001ae7d02000 (115558326272)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 8 ide-test /x86_64/ide/bmdma/no_busmaster
PASS 5 test-thread-pool /thread-pool/cancel
PASS 9 ide-test /x86_64/ide/flush/nodev
==9823==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 10 ide-test /x86_64/ide/flush/empty_drive
==9828==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 6 test-thread-pool /thread-pool/cancel-async
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-hbitmap -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-hbitmap"
PASS 1 test-hbitmap /hbitmap/granularity
---
PASS 4 test-hbitmap /hbitmap/iter/empty
PASS 11 ide-test /x86_64/ide/flush/retry_pci
PASS 5 test-hbitmap /hbitmap/iter/partial
==9840==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 6 test-hbitmap /hbitmap/iter/granularity
PASS 7 test-hbitmap /hbitmap/iter/iter_and_reset
PASS 12 ide-test /x86_64/ide/flush/retry_isa
---
PASS 14 test-hbitmap /hbitmap/set/twice
PASS 15 test-hbitmap /hbitmap/set/overlap
PASS 16 test-hbitmap /hbitmap/reset/empty
==9846==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 13 ide-test /x86_64/ide/cdrom/pio
PASS 17 test-hbitmap /hbitmap/reset/general
PASS 18 test-hbitmap /hbitmap/reset/all
---
PASS 28 test-hbitmap /hbitmap/truncate/shrink/medium
PASS 29 test-hbitmap /hbitmap/truncate/shrink/large
PASS 30 test-hbitmap /hbitmap/meta/zero
==9852==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 14 ide-test /x86_64/ide/cdrom/pio_large
==9858==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 15 ide-test /x86_64/ide/cdrom/dma
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/ahci-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="ahci-test"
==9872==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 ahci-test /x86_64/ahci/sanity
==9878==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 ahci-test /x86_64/ahci/pci_spec
==9884==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 31 test-hbitmap /hbitmap/meta/one
PASS 32 test-hbitmap /hbitmap/meta/byte
PASS 33 test-hbitmap /hbitmap/meta/word
PASS 3 ahci-test /x86_64/ahci/pci_enable
==9890==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 34 test-hbitmap /hbitmap/meta/sector
PASS 35 test-hbitmap /hbitmap/serialize/align
PASS 4 ahci-test /x86_64/ahci/hba_spec
==9896==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 5 ahci-test /x86_64/ahci/hba_enable
==9902==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 36 test-hbitmap /hbitmap/serialize/basic
PASS 37 test-hbitmap /hbitmap/serialize/part
PASS 38 test-hbitmap /hbitmap/serialize/zeroes
---
PASS 42 test-hbitmap /hbitmap/next_dirty_area/next_dirty_area_1
PASS 43 test-hbitmap /hbitmap/next_dirty_area/next_dirty_area_4
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-bdrv-drain -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-bdrv-drain"
==9911==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-bdrv-drain /bdrv-drain/nested
PASS 2 test-bdrv-drain /bdrv-drain/multiparent
==9908==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 test-bdrv-drain /bdrv-drain/set_aio_context
PASS 4 test-bdrv-drain /bdrv-drain/driver-cb/drain_all
PASS 5 test-bdrv-drain /bdrv-drain/driver-cb/drain
---
PASS 38 test-bdrv-drain /bdrv-drain/detach/driver_cb
PASS 39 test-bdrv-drain /bdrv-drain/attach/drain
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-bdrv-graph-mod -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-bdrv-graph-mod"
==9957==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-bdrv-graph-mod /bdrv-graph-mod/update-perm-tree
PASS 2 test-bdrv-graph-mod /bdrv-graph-mod/should-update-child
==9953==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-blockjob -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-blockjob"
==9966==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-blockjob /blockjob/ids
PASS 2 test-blockjob /blockjob/cancel/created
PASS 3 test-blockjob /blockjob/cancel/running
---
PASS 7 test-blockjob /blockjob/cancel/pending
PASS 8 test-blockjob /blockjob/cancel/concluded
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-blockjob-txn -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-blockjob-txn"
==9971==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-blockjob-txn /single/success
PASS 2 test-blockjob-txn /single/failure
PASS 3 test-blockjob-txn /single/cancel
---
PASS 6 test-blockjob-txn /pair/cancel
PASS 7 test-blockjob-txn /pair/fail-cancel-race
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-block-backend -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-block-backend"
==9976==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-block-backend /block-backend/drain_aio_error
PASS 2 test-block-backend /block-backend/drain_all_aio_error
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-block-iothread -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-block-iothread"
PASS 8 ahci-test /x86_64/ahci/reset
==9982==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-block-iothread /sync-op/pread
PASS 2 test-block-iothread /sync-op/pwrite
PASS 3 test-block-iothread /sync-op/load_vmstate
---
PASS 15 test-block-iothread /propagate/diamond
PASS 16 test-block-iothread /propagate/mirror
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-image-locking -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-image-locking"
==10004==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-image-locking /image-locking/basic
PASS 2 test-image-locking /image-locking/set-perm-abort
==9984==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-x86-cpuid -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-x86-cpuid"
PASS 1 test-x86-cpuid /cpuid/topology/basic
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-xbzrle -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-xbzrle"
==9984==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffeef9d0000; bottom 0x7fdaae9fe000; size: 0x002440fd2000 (155709153280)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 1 test-xbzrle /xbzrle/uleb
---
PASS 4 test-xbzrle /xbzrle/encode_decode_1_byte
PASS 5 test-xbzrle /xbzrle/encode_decode_overflow
PASS 9 ahci-test /x86_64/ahci/io/pio/lba28/simple/zero
==10020==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 6 test-xbzrle /xbzrle/encode_decode
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-vmstate -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-vmstate"
==10020==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffeb7c53000; bottom 0x7f52149fe000; size: 0x00aca3255000 (741471506432)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 1 test-vmstate /vmstate/tmp_struct
---
PASS 1 test-mul64 /host-utils/mulu64
PASS 2 test-mul64 /host-utils/muls64
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-int128 -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-int128"
==10036==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-int128 /int128/int128_and
PASS 2 test-int128 /int128/int128_add
PASS 3 test-int128 /int128/int128_sub
---
PASS 9 test-int128 /int128/int128_gt
PASS 10 test-int128 /int128/int128_rshift
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/rcutorture -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="rcutorture"
==10036==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffd36cb0000; bottom 0x7f95869fe000; size: 0x0067b02b2000 (445337247744)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 11 ahci-test /x86_64/ahci/io/pio/lba28/simple/high
==10065==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 rcutorture /rcu/torture/1reader
==10065==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fffc7f70000; bottom 0x7ff2077fe000; size: 0x000dc0772000 (59063607296)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 12 ahci-test /x86_64/ahci/io/pio/lba28/double/zero
==10087==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10087==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffd9fdfe000; bottom 0x7f6e879fe000; size: 0x008f18400000 (614587170816)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 2 rcutorture /rcu/torture/10readers
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-rcu-list -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-rcu-list"
PASS 13 ahci-test /x86_64/ahci/io/pio/lba28/double/low
==10100==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10100==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffc51a1d000; bottom 0x7f0a40ffe000; size: 0x00f210a1f000 (1039661133824)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 1 test-rcu-list /rcu/qlist/single-threaded
PASS 14 ahci-test /x86_64/ahci/io/pio/lba28/double/high
==10112==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10112==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fff158e0000; bottom 0x7f3d81ffe000; size: 0x00c1938e2000 (831404253184)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 2 test-rcu-list /rcu/qlist/short-few
PASS 15 ahci-test /x86_64/ahci/io/pio/lba28/long/zero
==10139==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10139==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fffbf09e000; bottom 0x7f6c21b7c000; size: 0x00939d522000 (633999597568)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 3 test-rcu-list /rcu/qlist/long-many
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-rcu-simpleq -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-rcu-simpleq"
PASS 1 test-rcu-simpleq /rcu/qsimpleq/single-threaded
PASS 16 ahci-test /x86_64/ahci/io/pio/lba28/long/low
==10158==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 test-rcu-simpleq /rcu/qsimpleq/short-few
==10158==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffe01b22000; bottom 0x7fa2df724000; size: 0x005b223fe000 (391416635392)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 17 ahci-test /x86_64/ahci/io/pio/lba28/long/high
==10185==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 18 ahci-test /x86_64/ahci/io/pio/lba28/short/zero
PASS 3 test-rcu-simpleq /rcu/qsimpleq/long-many
==10191==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-rcu-tailq -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-rcu-tailq"
PASS 19 ahci-test /x86_64/ahci/io/pio/lba28/short/low
==10204==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-rcu-tailq /rcu/qtailq/single-threaded
PASS 20 ahci-test /x86_64/ahci/io/pio/lba28/short/high
==10216==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10216==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffd72266000; bottom 0x7f69601fe000; size: 0x009412068000 (635957575680)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 2 test-rcu-tailq /rcu/qtailq/short-few
PASS 21 ahci-test /x86_64/ahci/io/pio/lba48/simple/zero
==10243==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10243==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffc76ea2000; bottom 0x7fa6715fe000; size: 0x0056058a4000 (369460133888)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 22 ahci-test /x86_64/ahci/io/pio/lba48/simple/low
==10249==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10249==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fffd9590000; bottom 0x7ffbaabfe000; size: 0x00042e992000 (17961656320)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 3 test-rcu-tailq /rcu/qtailq/long-many
---
PASS 7 test-qdist /qdist/binning/expand
PASS 8 test-qdist /qdist/binning/shrink
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-qht -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-qht"
==10256==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10256==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffeea6d8000; bottom 0x7fa989bfe000; size: 0x005560ada000 (366694211584)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 24 ahci-test /x86_64/ahci/io/pio/lba48/double/zero
==10270==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10270==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffde38b8000; bottom 0x7fad91bfe000; size: 0x005051cba000 (344969682944)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 25 ahci-test /x86_64/ahci/io/pio/lba48/double/low
==10276==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10276==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffc76d88000; bottom 0x7fb6f4dfe000; size: 0x004581f8a000 (298533298176)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 26 ahci-test /x86_64/ahci/io/pio/lba48/double/high
==10282==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10282==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fff42d80000; bottom 0x7f9ad69fe000; size: 0x00646c382000 (431312347136)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 27 ahci-test /x86_64/ahci/io/pio/lba48/long/zero
==10288==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10288==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fff2eeab000; bottom 0x7f9d2817c000; size: 0x006206d2f000 (421021282304)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 28 ahci-test /x86_64/ahci/io/pio/lba48/long/low
==10294==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10294==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7ffff1618000; bottom 0x7f2e7f7fe000; size: 0x00d171e1a000 (899558776832)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 29 ahci-test /x86_64/ahci/io/pio/lba48/long/high
==10300==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 30 ahci-test /x86_64/ahci/io/pio/lba48/short/zero
==10306==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 31 ahci-test /x86_64/ahci/io/pio/lba48/short/low
==10312==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 32 ahci-test /x86_64/ahci/io/pio/lba48/short/high
==10318==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 33 ahci-test /x86_64/ahci/io/dma/lba28/fragmented
==10324==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 34 ahci-test /x86_64/ahci/io/dma/lba28/retry
==10330==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 35 ahci-test /x86_64/ahci/io/dma/lba28/simple/zero
==10336==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 36 ahci-test /x86_64/ahci/io/dma/lba28/simple/low
==10342==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-qht /qht/mode/default
PASS 37 ahci-test /x86_64/ahci/io/dma/lba28/simple/high
PASS 2 test-qht /qht/mode/resize
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-qht-par -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-qht-par"
==10348==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 38 ahci-test /x86_64/ahci/io/dma/lba28/double/zero
PASS 1 test-qht-par /qht/parallel/2threads-0%updates-1s
==10364==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 test-qht-par /qht/parallel/2threads-20%updates-1s
PASS 39 ahci-test /x86_64/ahci/io/dma/lba28/double/low
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-bitops -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-bitops"
---
PASS 3 test-bitcnt /bitcnt/ctpop32
PASS 4 test-bitcnt /bitcnt/ctpop64
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-qdev-global-props -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-qdev-global-props"
==10379==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-qdev-global-props /qdev/properties/static/default
PASS 2 test-qdev-global-props /qdev/properties/static/global
PASS 3 test-qdev-global-props /qdev/properties/dynamic/global
---
PASS 9 test-keyval /keyval/visit/alternate
PASS 10 test-keyval /keyval/visit/any
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-write-threshold -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-write-threshold"
==10415==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-write-threshold /write-threshold/not-set-on-init
PASS 2 test-write-threshold /write-threshold/set-get
PASS 3 test-write-threshold /write-threshold/multi-set-get
---
PASS 15 test-crypto-secret /crypto/secret/crypt/missingiv
PASS 16 test-crypto-secret /crypto/secret/crypt/badiv
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-crypto-tlscredsx509 -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-crypto-tlscredsx509"
==10441==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/perfectserver
PASS 42 ahci-test /x86_64/ahci/io/dma/lba28/long/low
PASS 2 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/perfectclient
==10461==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodca1
PASS 43 ahci-test /x86_64/ahci/io/dma/lba28/long/high
==10467==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 44 ahci-test /x86_64/ahci/io/dma/lba28/short/zero
==10473==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 4 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodca2
PASS 45 ahci-test /x86_64/ahci/io/dma/lba28/short/low
PASS 5 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodca3
PASS 6 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/badca1
PASS 7 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/badca2
PASS 8 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/badca3
==10479==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 9 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver1
PASS 46 ahci-test /x86_64/ahci/io/dma/lba28/short/high
PASS 10 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver2
==10485==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 11 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver3
PASS 47 ahci-test /x86_64/ahci/io/dma/lba48/simple/zero
==10491==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 12 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver4
PASS 13 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver5
PASS 48 ahci-test /x86_64/ahci/io/dma/lba48/simple/low
PASS 14 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver6
==10497==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 15 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/goodserver7
PASS 16 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/badserver1
PASS 17 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/badserver2
---
PASS 37 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/missingca
PASS 38 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/missingserver
PASS 39 test-crypto-tlscredsx509 /qcrypto/tlscredsx509/missingclient
==10503==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-crypto-tlssession -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-crypto-tlssession"
PASS 50 ahci-test /x86_64/ahci/io/dma/lba48/double/zero
PASS 1 test-crypto-tlssession /qcrypto/tlssession/psk
==10514==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 test-crypto-tlssession /qcrypto/tlssession/basicca
PASS 3 test-crypto-tlssession /qcrypto/tlssession/differentca
PASS 4 test-crypto-tlssession /qcrypto/tlssession/altname1
PASS 5 test-crypto-tlssession /qcrypto/tlssession/altname2
PASS 51 ahci-test /x86_64/ahci/io/dma/lba48/double/low
PASS 6 test-crypto-tlssession /qcrypto/tlssession/altname3
==10520==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 7 test-crypto-tlssession /qcrypto/tlssession/altname4
PASS 52 ahci-test /x86_64/ahci/io/dma/lba48/double/high
==10526==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 8 test-crypto-tlssession /qcrypto/tlssession/altname5
PASS 9 test-crypto-tlssession /qcrypto/tlssession/altname6
PASS 53 ahci-test /x86_64/ahci/io/dma/lba48/long/zero
PASS 10 test-crypto-tlssession /qcrypto/tlssession/wildcard1
==10532==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 11 test-crypto-tlssession /qcrypto/tlssession/wildcard2
PASS 12 test-crypto-tlssession /qcrypto/tlssession/wildcard3
PASS 13 test-crypto-tlssession /qcrypto/tlssession/wildcard4
PASS 14 test-crypto-tlssession /qcrypto/tlssession/wildcard5
PASS 54 ahci-test /x86_64/ahci/io/dma/lba48/long/low
==10538==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 15 test-crypto-tlssession /qcrypto/tlssession/wildcard6
PASS 55 ahci-test /x86_64/ahci/io/dma/lba48/long/high
PASS 16 test-crypto-tlssession /qcrypto/tlssession/cachain
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-qga -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-qga"
==10544==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 56 ahci-test /x86_64/ahci/io/dma/lba48/short/zero
PASS 1 test-qga /qga/sync-delimited
PASS 2 test-qga /qga/sync
---
PASS 15 test-qga /qga/invalid-cmd
PASS 16 test-qga /qga/invalid-args
PASS 17 test-qga /qga/fsfreeze-status
==10556==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 18 test-qga /qga/blacklist
PASS 19 test-qga /qga/config
PASS 57 ahci-test /x86_64/ahci/io/dma/lba48/short/low
PASS 20 test-qga /qga/guest-exec
PASS 21 test-qga /qga/guest-exec-invalid
==10567==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 22 test-qga /qga/guest-get-osinfo
PASS 23 test-qga /qga/guest-get-host-name
PASS 24 test-qga /qga/guest-get-timezone
---
PASS 7 test-util-sockets /socket/fd-pass/num/bad
PASS 8 test-util-sockets /socket/fd-pass/num/nocli
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-authz-simple -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-authz-simple"
==10589==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-authz-simple /authz/simple
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-authz-list -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-authz-list"
PASS 1 test-authz-list /auth/list/complex
---
PASS 4 test-io-channel-file /io/channel/pipe/sync
PASS 5 test-io-channel-file /io/channel/pipe/async
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-io-channel-tls -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-io-channel-tls"
==10670==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-io-channel-tls /qio/channel/tls/basic
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-io-channel-command -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-io-channel-command"
PASS 1 test-io-channel-command /io/channel/command/fifo/sync
---
PASS 17 test-crypto-pbkdf /crypto/pbkdf/nonrfc/sha384/iter1200
PASS 18 test-crypto-pbkdf /crypto/pbkdf/nonrfc/ripemd160/iter1200
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-crypto-ivgen -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-crypto-ivgen"
==10700==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-crypto-ivgen /crypto/ivgen/plain/1
PASS 2 test-crypto-ivgen /crypto/ivgen/plain/1f2e3d4c
PASS 3 test-crypto-ivgen /crypto/ivgen/plain/1f2e3d4c5b6a7988
---
PASS 2 test-logging /logging/parse_path
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-replication -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-replication"
PASS 61 ahci-test /x86_64/ahci/flush/simple
==10737==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 test-replication /replication/primary/read
==10739==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 test-replication /replication/primary/write
PASS 62 ahci-test /x86_64/ahci/flush/retry
PASS 3 test-replication /replication/primary/start
---
PASS 5 test-replication /replication/primary/do_checkpoint
PASS 6 test-replication /replication/primary/get_error_all
PASS 7 test-replication /replication/secondary/read
==10747==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 8 test-replication /replication/secondary/write
==10753==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10737==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fff08a08000; bottom 0x7f0548afc000; size: 0x00f9bff0c000 (1072667082752)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 63 ahci-test /x86_64/ahci/flush/migrate
PASS 9 test-replication /replication/secondary/start
==10786==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10791==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 10 test-replication /replication/secondary/stop
PASS 64 ahci-test /x86_64/ahci/migrate/sanity
==10800==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10805==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 11 test-replication /replication/secondary/do_checkpoint
PASS 12 test-replication /replication/secondary/get_error_all
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} tests/test-bufferiszero -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="test-bufferiszero"
PASS 65 ahci-test /x86_64/ahci/migrate/dma/simple
==10818==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10823==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 66 ahci-test /x86_64/ahci/migrate/dma/halted
==10832==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10837==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 67 ahci-test /x86_64/ahci/migrate/ncq/simple
==10847==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10852==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 68 ahci-test /x86_64/ahci/migrate/ncq/halted
==10861==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 69 ahci-test /x86_64/ahci/cdrom/eject
==10866==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 70 ahci-test /x86_64/ahci/cdrom/dma/single
==10872==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 71 ahci-test /x86_64/ahci/cdrom/dma/multi
==10878==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 72 ahci-test /x86_64/ahci/cdrom/pio/single
==10884==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
==10884==WARNING: ASan is ignoring requested __asan_handle_no_return: stack top: 0x7fff61b09000; bottom 0x7f6b8bb44000; size: 0x0093d5fc5000 (634950275072)
False positive error reports may follow
For details see https://github.com/google/sanitizers/issues/189
PASS 73 ahci-test /x86_64/ahci/cdrom/pio/multi
==10890==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 74 ahci-test /x86_64/ahci/cdrom/pio/bcl
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/hd-geo-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="hd-geo-test"
PASS 1 hd-geo-test /x86_64/hd-geo/ide/none
==10904==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 hd-geo-test /x86_64/hd-geo/ide/drive/cd_0
==10910==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 hd-geo-test /x86_64/hd-geo/ide/drive/mbr/blank
==10916==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 4 hd-geo-test /x86_64/hd-geo/ide/drive/mbr/lba
==10922==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 5 hd-geo-test /x86_64/hd-geo/ide/drive/mbr/chs
==10928==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 6 hd-geo-test /x86_64/hd-geo/ide/device/mbr/blank
==10934==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 7 hd-geo-test /x86_64/hd-geo/ide/device/mbr/lba
==10940==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 8 hd-geo-test /x86_64/hd-geo/ide/device/mbr/chs
==10946==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 9 hd-geo-test /x86_64/hd-geo/ide/device/user/chs
==10951==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 10 hd-geo-test /x86_64/hd-geo/ide/device/user/chst
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/boot-order-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="boot-order-test"
PASS 1 test-bufferiszero /cutils/bufferiszero
---
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11036==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 1 bios-tables-test /x86_64/acpi/piix4
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11042==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 2 bios-tables-test /x86_64/acpi/q35
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11048==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 bios-tables-test /x86_64/acpi/piix4/bridge
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11054==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 4 bios-tables-test /x86_64/acpi/piix4/ipmi
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11060==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 5 bios-tables-test /x86_64/acpi/piix4/cpuhp
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11067==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 6 bios-tables-test /x86_64/acpi/piix4/memhp
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11073==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 7 bios-tables-test /x86_64/acpi/piix4/numamem
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11079==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 8 bios-tables-test /x86_64/acpi/piix4/dimmpxm
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11088==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 9 bios-tables-test /x86_64/acpi/q35/bridge
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11094==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 10 bios-tables-test /x86_64/acpi/q35/mmio64
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11100==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 11 bios-tables-test /x86_64/acpi/q35/ipmi
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11106==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 12 bios-tables-test /x86_64/acpi/q35/cpuhp
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11113==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 13 bios-tables-test /x86_64/acpi/q35/memhp
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11119==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 14 bios-tables-test /x86_64/acpi/q35/numamem
Could not access KVM kernel module: No such file or directory
qemu-system-x86_64: failed to initialize KVM: No such file or directory
qemu-system-x86_64: Back to tcg accelerator
==11125==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 15 bios-tables-test /x86_64/acpi/q35/dimmpxm
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/boot-serial-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="boot-serial-test"
PASS 1 boot-serial-test /x86_64/boot-serial/isapc
---
PASS 1 i440fx-test /x86_64/i440fx/defaults
PASS 2 i440fx-test /x86_64/i440fx/pam
PASS 3 i440fx-test /x86_64/i440fx/firmware/bios
==11209==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 4 i440fx-test /x86_64/i440fx/firmware/pflash
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/fw_cfg-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="fw_cfg-test"
PASS 1 fw_cfg-test /x86_64/fw_cfg/signature
---
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/drive_del-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="drive_del-test"
PASS 1 drive_del-test /x86_64/drive_del/without-dev
PASS 2 drive_del-test /x86_64/drive_del/after_failed_device_add
==11297==WARNING: ASan doesn't fully support makecontext/swapcontext functions and may produce false positives in some cases!
PASS 3 drive_del-test /x86_64/blockdev/drive_del_device_del
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/wdt_ib700-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="wdt_ib700-test"
PASS 1 wdt_ib700-test /x86_64/wdt_ib700/pause
---
MALLOC_PERTURB_=${MALLOC_PERTURB_:-$(( ${RANDOM:-0} % 255 + 1))} QTEST_QEMU_BINARY=x86_64-softmmu/qemu-system-x86_64 QTEST_QEMU_IMG=qemu-img tests/intel-hda-test -m=quick -k --tap < /dev/null | ./scripts/tap-driver.pl --test-name="intel-hda-test"
=================================================================
==11432==ERROR: LeakSanitizer: detected memory leaks
Direct leak of 48 byte(s) in 1 object(s) allocated from:
#0 0x559fe8f96b6e in calloc (/tmp/qemu-test/build/x86_64-softmmu/qemu-system-x86_64+0x1a0bb6e)
---
SUMMARY: AddressSanitizer: 48 byte(s) leaked in 1 allocation(s).
/tmp/qemu-test/src/tests/libqtest.c:137: kill_qemu() tried to terminate QEMU process but encountered exit status 1
ERROR - too few tests run (expected 2, got 0)
make: *** [/tmp/qemu-test/src/tests/Makefile.include:897: check-qtest-x86_64] Error 1
make: *** Waiting for unfinished jobs....
Traceback (most recent call last):
The full log is available at
http://patchew.org/logs/cover.1562695780.git.DirtY.iCE.hu@gmail.com/testing.asan/?type=message.
---
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Please send your feedback to patchew-devel@redhat.com
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