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[Qemu-devel] [PULL 07/20] alsaaudio: do not use global variables
From: |
Gerd Hoffmann |
Subject: |
[Qemu-devel] [PULL 07/20] alsaaudio: do not use global variables |
Date: |
Mon, 15 Jun 2015 14:27:58 +0200 |
From: Kővágó, Zoltán <address@hidden>
Signed-off-by: Kővágó, Zoltán <address@hidden>
Signed-off-by: Gerd Hoffmann <address@hidden>
---
audio/alsaaudio.c | 160 +++++++++++++++++++++++++++++-------------------------
1 file changed, 87 insertions(+), 73 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5a5bb14..4bcf58f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -33,30 +33,7 @@
#define AUDIO_CAP "alsa"
#include "audio_int.h"
-struct pollhlp {
- snd_pcm_t *handle;
- struct pollfd *pfds;
- int count;
- int mask;
-};
-
-typedef struct ALSAVoiceOut {
- HWVoiceOut hw;
- int wpos;
- int pending;
- void *pcm_buf;
- snd_pcm_t *handle;
- struct pollhlp pollhlp;
-} ALSAVoiceOut;
-
-typedef struct ALSAVoiceIn {
- HWVoiceIn hw;
- snd_pcm_t *handle;
- void *pcm_buf;
- struct pollhlp pollhlp;
-} ALSAVoiceIn;
-
-static struct {
+typedef struct ALSAConf {
int size_in_usec_in;
int size_in_usec_out;
const char *pcm_name_in;
@@ -73,13 +50,32 @@ static struct {
int buffer_size_out_overridden;
int period_size_out_overridden;
int verbose;
-} conf = {
- .buffer_size_out = 4096,
- .period_size_out = 1024,
- .pcm_name_out = "default",
- .pcm_name_in = "default",
+} ALSAConf;
+
+struct pollhlp {
+ snd_pcm_t *handle;
+ struct pollfd *pfds;
+ ALSAConf *conf;
+ int count;
+ int mask;
};
+typedef struct ALSAVoiceOut {
+ HWVoiceOut hw;
+ int wpos;
+ int pending;
+ void *pcm_buf;
+ snd_pcm_t *handle;
+ struct pollhlp pollhlp;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+ HWVoiceIn hw;
+ snd_pcm_t *handle;
+ void *pcm_buf;
+ struct pollhlp pollhlp;
+} ALSAVoiceIn;
+
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
@@ -184,6 +180,7 @@ static void alsa_poll_handler (void *opaque)
snd_pcm_state_t state;
struct pollhlp *hlp = opaque;
unsigned short revents;
+ ALSAConf *conf = hlp->conf;
count = poll (hlp->pfds, hlp->count, 0);
if (count < 0) {
@@ -205,7 +202,7 @@ static void alsa_poll_handler (void *opaque)
}
if (!(revents & hlp->mask)) {
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("revents = %d\n", revents);
}
return;
@@ -242,6 +239,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct
pollhlp *hlp, int mask)
{
int i, count, err;
struct pollfd *pfds;
+ ALSAConf *conf = hlp->conf;
count = snd_pcm_poll_descriptors_count (handle);
if (count <= 0) {
@@ -269,12 +267,12 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct
pollhlp *hlp, int mask)
qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
}
if (pfds[i].events & POLLOUT) {
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
}
qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
pfds[i].events, i, pfds[i].fd, err);
}
@@ -464,14 +462,15 @@ static void alsa_set_threshold (snd_pcm_t *handle,
snd_pcm_uframes_t threshold)
}
static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep)
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ ALSAConf *conf)
{
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+ const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
@@ -510,7 +509,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
- if (err < 0 && conf.verbose) {
+ if (err < 0 && conf->verbose) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
}
@@ -642,7 +641,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf.threshold) {
+ if (!in && conf->threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
@@ -664,7 +663,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
break;
}
- threshold = (conf.threshold * bytes_per_sec) / 1000;
+ threshold = (conf->threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
@@ -674,7 +673,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
*handlep = handle;
- if (conf.verbose &&
+ if (conf->verbose &&
(obtfmt != req->fmt ||
obt->nchannels != req->nchannels ||
obt->freq != req->freq)) {
@@ -717,6 +716,7 @@ static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
static void alsa_write_pending (ALSAVoiceOut *alsa)
{
HWVoiceOut *hw = &alsa->hw;
+ ALSAConf *conf = alsa->pollhlp.conf;
while (alsa->pending) {
int left_till_end_samples = hw->samples - alsa->wpos;
@@ -731,7 +731,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
if (written <= 0) {
switch (written) {
case 0:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Failed to write %d frames (wrote zero)\n",
len);
}
return;
@@ -742,7 +742,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
len);
return;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Recovering from playback xrun\n");
}
continue;
@@ -755,7 +755,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
len);
return;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Resuming suspended output stream\n");
}
continue;
@@ -815,18 +815,19 @@ static int alsa_init_out(HWVoiceOut *hw, struct
audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
+ ALSAConf *conf = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_out;
- req.buffer_size = conf.buffer_size_out;
- req.size_in_usec = conf.size_in_usec_out;
+ req.period_size = conf->period_size_out;
+ req.buffer_size = conf->buffer_size_out;
+ req.size_in_usec = conf->size_in_usec_out;
req.override_mask =
- (conf.period_size_out_overridden ? 1 : 0) |
- (conf.buffer_size_out_overridden ? 2 : 0);
+ (conf->period_size_out_overridden ? 1 : 0) |
+ (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle)) {
+ if (alsa_open (0, &req, &obt, &handle, conf)) {
return -1;
}
@@ -847,6 +848,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as,
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
@@ -924,18 +926,19 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
+ ALSAConf *conf = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf.period_size_in;
- req.buffer_size = conf.buffer_size_in;
- req.size_in_usec = conf.size_in_usec_in;
+ req.period_size = conf->period_size_in;
+ req.buffer_size = conf->buffer_size_in;
+ req.size_in_usec = conf->size_in_usec_in;
req.override_mask =
- (conf.period_size_in_overridden ? 1 : 0) |
- (conf.buffer_size_in_overridden ? 2 : 0);
+ (conf->period_size_in_overridden ? 1 : 0) |
+ (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle)) {
+ if (alsa_open (1, &req, &obt, &handle, conf)) {
return -1;
}
@@ -956,6 +959,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
}
alsa->handle = handle;
+ alsa->pollhlp.conf = conf;
return 0;
}
@@ -986,6 +990,7 @@ static int alsa_run_in (HWVoiceIn *hw)
};
snd_pcm_sframes_t avail;
snd_pcm_uframes_t read_samples = 0;
+ ALSAConf *conf = alsa->pollhlp.conf;
if (!dead) {
return 0;
@@ -1011,12 +1016,12 @@ static int alsa_run_in (HWVoiceIn *hw)
dolog ("Failed to resume suspended input stream\n");
return 0;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Resuming suspended input stream\n");
}
break;
default:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("No frames available and ALSA state is %d\n", state);
}
return 0;
@@ -1053,7 +1058,7 @@ static int alsa_run_in (HWVoiceIn *hw)
if (nread <= 0) {
switch (nread) {
case 0:
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Failed to read %ld frames (read zero)\n", len);
}
goto exit;
@@ -1063,7 +1068,7 @@ static int alsa_run_in (HWVoiceIn *hw)
alsa_logerr (nread, "Failed to read %ld frames\n",
len);
goto exit;
}
- if (conf.verbose) {
+ if (conf->verbose) {
dolog ("Recovering from capture xrun\n");
}
continue;
@@ -1137,80 +1142,89 @@ static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
return -1;
}
+static ALSAConf glob_conf = {
+ .buffer_size_out = 4096,
+ .period_size_out = 1024,
+ .pcm_name_out = "default",
+ .pcm_name_in = "default",
+};
+
static void *alsa_audio_init (void)
{
- return &conf;
+ ALSAConf *conf = g_malloc(sizeof(ALSAConf));
+ *conf = glob_conf;
+ return conf;
}
static void alsa_audio_fini (void *opaque)
{
- (void) opaque;
+ g_free(opaque);
}
static struct audio_option alsa_options[] = {
{
.name = "DAC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_out,
+ .valp = &glob_conf.size_in_usec_out,
.descr = "DAC period/buffer size in microseconds (otherwise in
frames)"
},
{
.name = "DAC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_out,
+ .valp = &glob_conf.period_size_out,
.descr = "DAC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_out_overridden
+ .overriddenp = &glob_conf.period_size_out_overridden
},
{
.name = "DAC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_out,
+ .valp = &glob_conf.buffer_size_out,
.descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_out_overridden
+ .overriddenp = &glob_conf.buffer_size_out_overridden
},
{
.name = "ADC_SIZE_IN_USEC",
.tag = AUD_OPT_BOOL,
- .valp = &conf.size_in_usec_in,
+ .valp = &glob_conf.size_in_usec_in,
.descr =
"ADC period/buffer size in microseconds (otherwise in frames)"
},
{
.name = "ADC_PERIOD_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.period_size_in,
+ .valp = &glob_conf.period_size_in,
.descr = "ADC period size (0 to go with system default)",
- .overriddenp = &conf.period_size_in_overridden
+ .overriddenp = &glob_conf.period_size_in_overridden
},
{
.name = "ADC_BUFFER_SIZE",
.tag = AUD_OPT_INT,
- .valp = &conf.buffer_size_in,
+ .valp = &glob_conf.buffer_size_in,
.descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &conf.buffer_size_in_overridden
+ .overriddenp = &glob_conf.buffer_size_in_overridden
},
{
.name = "THRESHOLD",
.tag = AUD_OPT_INT,
- .valp = &conf.threshold,
+ .valp = &glob_conf.threshold,
.descr = "(undocumented)"
},
{
.name = "DAC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_out,
+ .valp = &glob_conf.pcm_name_out,
.descr = "DAC device name (for instance dmix)"
},
{
.name = "ADC_DEV",
.tag = AUD_OPT_STR,
- .valp = &conf.pcm_name_in,
+ .valp = &glob_conf.pcm_name_in,
.descr = "ADC device name"
},
{
.name = "VERBOSE",
.tag = AUD_OPT_BOOL,
- .valp = &conf.verbose,
+ .valp = &glob_conf.verbose,
.descr = "Behave in a more verbose way"
},
{ /* End of list */ }
--
1.8.3.1
- [Qemu-devel] [PULL 12/20] coreaudio: do not use global variables where possible, (continued)
- [Qemu-devel] [PULL 12/20] coreaudio: do not use global variables where possible, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 15/20] audio: remove LOG_TO_MONITOR along with default_mon, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 03/20] audio: remove winwave audio driver, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 08/20] ossaudio: do not use global variables, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 16/20] audio: remove plive, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 06/20] paaudio: do not use global variables, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 02/20] audio: remove fmod backend, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 01/20] audio: remove esd backend, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 05/20] audio: expose drv_opaque to init_out and init_in, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 19/20] alsaaudio: use trace events instead of verbose, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 07/20] alsaaudio: do not use global variables,
Gerd Hoffmann <=
- [Qemu-devel] [PULL 17/20] dsoundaudio: remove *_retries kludges, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 20/20] ossaudio: use trace events instead of debug config flag, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 18/20] dsoundaudio: remove primary buffer, Gerd Hoffmann, 2015/06/15
- [Qemu-devel] [PULL 11/20] dsoundaudio: do not use global variables, Gerd Hoffmann, 2015/06/15
- Re: [Qemu-devel] [PULL 00/20] audio patch queue, Peter Maydell, 2015/06/15