Audio backend cleanup

index fd019a0..9dd6b7f 100644
@@ -210,7 +210,6 @@ extern struct audio_driver fmod_audio_driver;
 extern struct audio_driver alsa_audio_driver;
 extern struct audio_driver coreaudio_audio_driver;
 extern struct audio_driver dsound_audio_driver;
-extern struct audio_driver esd_audio_driver;
 extern struct audio_driver pa_audio_driver;
 extern struct audio_driver spice_audio_driver;
 extern struct audio_driver winwave_audio_driver;

557 lines not on this page · raw message

index 9dd6b7f..7445602 100644
@@ -206,7 +206,6 @@ extern struct audio_driver no_audio_driver;
 extern struct audio_driver oss_audio_driver;
 extern struct audio_driver sdl_audio_driver;
 extern struct audio_driver wav_audio_driver;
-extern struct audio_driver fmod_audio_driver;
 extern struct audio_driver alsa_audio_driver;
 extern struct audio_driver coreaudio_audio_driver;
 extern struct audio_driver dsound_audio_driver;

685 lines not on this page · raw message

index 7445602..d593879 100644
@@ -204,7 +204,6 @@ struct AudioState {
 
 extern struct audio_driver no_audio_driver;
 extern struct audio_driver oss_audio_driver;
-extern struct audio_driver sdl_audio_driver;
 extern struct audio_driver wav_audio_driver;
 extern struct audio_driver alsa_audio_driver;
 extern struct audio_driver coreaudio_audio_driver;

458 lines not on this page · raw message

index d593879..92a2f3c 100644
@@ -210,7 +210,6 @@ extern struct audio_driver coreaudio_audio_driver;
 extern struct audio_driver dsound_audio_driver;
 extern struct audio_driver pa_audio_driver;
 extern struct audio_driver spice_audio_driver;
-extern struct audio_driver winwave_audio_driver;
 extern const struct mixeng_volume nominal_volume;
 
 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);

717 lines not on this page · raw message

index 74ead97..eb33bff 100644
@@ -819,7 +819,8 @@ static void alsa_fini_out (HWVoiceOut *hw)
     alsa->pcm_buf = NULL;
 }
 
-static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
+                         void *drv_opaque)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
     struct alsa_params_req req;

@@ -827,6 +828,8 @@ static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
     snd_pcm_t *handle;
     struct audsettings obt_as;
 
+    (void) drv_opaque;
+
     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;

@@ -928,7 +931,7 @@ static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return -1;
 }
 
-static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
     struct alsa_params_req req;

@@ -936,6 +939,8 @@ static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
     snd_pcm_t *handle;
     struct audsettings obt_as;
 
+    (void) drv_opaque;
+
     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;

index 92a2f3c..a3efe7b 100644
@@ -156,13 +156,13 @@ struct audio_driver {
 };
 
 struct audio_pcm_ops {
-    int  (*init_out)(HWVoiceOut *hw, struct audsettings *as);
+    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);
+    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);

index 584e536..f716d97 100644
@@ -262,7 +262,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
 #ifdef DAC
     QLIST_INIT (&hw->cap_head);
 #endif
-    if (glue (hw->pcm_ops->init_, TYPE) (hw, as)) {
+    if (glue (hw->pcm_ops->init_, TYPE) (hw, as, s->drv_opaque)) {
         goto err0;
     }
 

index 5964c62..f38fd82 100644
@@ -287,7 +287,8 @@ 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)
+static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
+                              void *drv_opaque)
 {
     OSStatus status;
     coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;

@@ -296,6 +297,8 @@ static int coreaudio_init_out (HWVoiceOut *hw, struct audsettings *as)
     const char *typ = "playback";
     AudioValueRange frameRange;
 
+    (void) drv_opaque;
+
     /* create mutex */
     err = pthread_mutex_init(&core->mutex, NULL);
     if (err) {

index 8b37d16..98276fb 100644
@@ -174,9 +174,11 @@ static void dsound_fini_out (HWVoiceOut *hw)
 }
 
 #ifdef DSBTYPE_IN
-static int dsound_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int dsound_init_in(HWVoiceIn *hw, struct audsettings *as,
+                          void *drv_opaque)
 #else
-static int dsound_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
+                           void *drv_opaque)
 #endif
 {
     int err;

index cb38662..dd1a87a 100644
@@ -63,8 +63,9 @@ 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)
+static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
 {
+    (void) drv_opaque;
     audio_pcm_init_info (&hw->info, as);
     hw->samples = 1024;
     return 0;

@@ -82,8 +83,9 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static int no_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
+    (void) drv_opaque;
     audio_pcm_init_info (&hw->info, as);
     hw->samples = 1024;
     return 0;

index 4db2ca6..80ac610 100644
@@ -510,7 +510,8 @@ static void oss_fini_out (HWVoiceOut *hw)
     }
 }
 
-static int oss_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
 {
     OSSVoiceOut *oss = (OSSVoiceOut *) hw;
     struct oss_params req, obt;

@@ -520,6 +521,8 @@ static int oss_init_out (HWVoiceOut *hw, struct audsettings *as)
     audfmt_e effective_fmt;
     struct audsettings obt_as;
 
+    (void) drv_opaque;
+
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);

@@ -676,7 +679,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static int oss_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
     OSSVoiceIn *oss = (OSSVoiceIn *) hw;
     struct oss_params req, obt;

@@ -686,6 +689,8 @@ static int oss_init_in (HWVoiceIn *hw, struct audsettings *as)
     audfmt_e effective_fmt;
     struct audsettings obt_as;
 
+    (void) drv_opaque;
+
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);

index 90ff245..11c2d29 100644
@@ -534,7 +534,8 @@ fail:
     return NULL;
 }
 
-static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
 {
     int error;
     static pa_sample_spec ss;

@@ -542,6 +543,8 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
     struct audsettings obt_as = *as;
     PAVoiceOut *pa = (PAVoiceOut *) hw;
 
+    (void) drv_opaque;
+
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
     ss.rate = as->freq;

@@ -601,13 +604,15 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
     return -1;
 }
 
-static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
+static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
     int error;
     static pa_sample_spec ss;
     struct audsettings obt_as = *as;
     PAVoiceIn *pa = (PAVoiceIn *) hw;
 
+    (void) drv_opaque;
+
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
     ss.rate = as->freq;

index 7b79bed..78ca04c 100644
@@ -115,11 +115,14 @@ static int rate_get_samples (struct audio_pcm_info *info, SpiceRateCtl *rate)
 
 /* playback */
 
-static int line_out_init (HWVoiceOut *hw, struct audsettings *as)
+static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
+                         void *drv_opaque)
 {
     SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
     struct audsettings settings;
 
+    (void) drv_opaque;
+
 #if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
     settings.freq       = spice_server_get_best_playback_rate(NULL);
 #else

@@ -243,11 +246,13 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
 
 /* record */
 
-static int line_in_init (HWVoiceIn *hw, struct audsettings *as)
+static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
     SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
     struct audsettings settings;
 
+    (void) drv_opaque;
+
 #if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
     settings.freq       = spice_server_get_best_record_rate(NULL);
 #else

index 6846a1a..5132aed 100644
@@ -105,7 +105,8 @@ static void le_store (uint8_t *buf, uint32_t val, int len)
     }
 }
 
-static int wav_init_out (HWVoiceOut *hw, struct audsettings *as)
+static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
+                        void *drv_opaque)
 {
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     int bits16 = 0, stereo = 0;

@@ -118,6 +119,7 @@ static int wav_init_out (HWVoiceOut *hw, struct audsettings *as)
     struct audsettings wav_as = conf.settings;
 
     (void) as;
+    (void) drv_opaque;
 
     stereo = wav_as.nchannels == 2;
     switch (wav_as.fmt) {

index eb33bff..d7e181b 100644
@@ -33,9 +33,29 @@
 #define AUDIO_CAP "alsa"
 #include "audio_int.h"
 
+typedef struct ALSAConf {
+    int size_in_usec_in;
+    int size_in_usec_out;
+    const char *pcm_name_in;
+    const char *pcm_name_out;
+    unsigned int buffer_size_in;
+    unsigned int period_size_in;
+    unsigned int buffer_size_out;
+    unsigned int period_size_out;
+    unsigned int threshold;
+
+    int buffer_size_in_overridden;
+    int period_size_in_overridden;
+
+    int buffer_size_out_overridden;
+    int period_size_out_overridden;
+    int verbose;
+} ALSAConf;
+
 struct pollhlp {
     snd_pcm_t *handle;
     struct pollfd *pfds;
+    ALSAConf *conf;
     int count;
     int mask;
 };

@@ -56,30 +76,6 @@ typedef struct ALSAVoiceIn {
     struct pollhlp pollhlp;
 } ALSAVoiceIn;
 
-static struct {
-    int size_in_usec_in;
-    int size_in_usec_out;
-    const char *pcm_name_in;
-    const char *pcm_name_out;
-    unsigned int buffer_size_in;
-    unsigned int period_size_in;
-    unsigned int buffer_size_out;
-    unsigned int period_size_out;
-    unsigned int threshold;
-
-    int buffer_size_in_overridden;
-    int period_size_in_overridden;
-
-    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",
-};
-
 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) {

@@ -270,13 +268,13 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
                                        NULL, hlp);
         }
         if (pfds[i].events & POLLOUT) {
-            if (conf.verbose) {
+            if (conf->verbose) {
                 dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
             }
             err = 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);
         }

@@ -476,14 +474,15 @@ 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;

@@ -522,7 +521,7 @@ 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);
     }
 

@@ -654,7 +653,7 @@ 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;
 

@@ -676,7 +675,7 @@ 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);
     }
 

@@ -686,7 +685,7 @@ 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)) {

@@ -729,6 +728,7 @@ 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;

@@ -743,7 +743,7 @@ 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;

@@ -754,7 +754,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
                                      len);
                         return;
                     }
-                    if (conf.verbose) {
+                    if (conf->verbose) {
                         dolog ("Recovering from playback xrun\n");
                     }
                     continue;

@@ -767,7 +767,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
                                      len);
                         return;
                     }
-                    if (conf.verbose) {
+                    if (conf->verbose) {
                         dolog ("Resuming suspended output stream\n");
                     }
                     continue;

@@ -827,20 +827,19 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
     struct alsa_params_obt obt;
     snd_pcm_t *handle;
     struct audsettings obt_as;
-
-    (void) drv_opaque;
+    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;
     }
 

@@ -861,6 +860,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     alsa->handle = handle;
+    alsa->pollhlp.conf = conf;
     return 0;
 }
 

@@ -938,20 +938,19 @@ 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;
-
-    (void) drv_opaque;
+    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;
     }
 

@@ -972,6 +971,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     alsa->handle = handle;
+    alsa->pollhlp.conf = conf;
     return 0;
 }
 

@@ -1002,6 +1002,7 @@ 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;

@@ -1027,12 +1028,12 @@ 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;

@@ -1069,7 +1070,7 @@ 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;

@@ -1079,7 +1080,7 @@ 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;

@@ -1153,80 +1154,89 @@ 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 */ }

index 11c2d29..35e8887 100644
@@ -9,6 +9,19 @@
 #include "audio_pt_int.h"
 
 typedef struct {
+    int samples;
+    char *server;
+    char *sink;
+    char *source;
+} PAConf;
+
+typedef struct {
+    PAConf conf;
+    pa_threaded_mainloop *mainloop;
+    pa_context *context;
+} paaudio;
+
+typedef struct {
     HWVoiceOut hw;
     int done;
     int live;

@@ -17,6 +30,7 @@ typedef struct {
     pa_stream *stream;
     void *pcm_buf;
     struct audio_pt pt;
+    paaudio *g;
 } PAVoiceOut;
 
 typedef struct {

@@ -30,21 +44,9 @@ typedef struct {
     struct audio_pt pt;
     const void *read_data;
     size_t read_index, read_length;
+    paaudio *g;
 } PAVoiceIn;
 
-typedef struct {
-    int samples;
-    char *server;
-    char *sink;
-    char *source;
-    pa_threaded_mainloop *mainloop;
-    pa_context *context;
-} paaudio;
-
-static paaudio glob_paaudio = {
-    .samples = 4096,
-};
-
 static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
 {
     va_list ap;

@@ -106,7 +108,7 @@ 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 = &glob_paaudio;
+    paaudio *g = p->g;
 
     pa_threaded_mainloop_lock (g->mainloop);
 

@@ -160,7 +162,7 @@ unlock_and_fail:
 
 static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
 {
-    paaudio *g = &glob_paaudio;
+    paaudio *g = p->g;
 
     pa_threaded_mainloop_lock (g->mainloop);
 

@@ -222,7 +224,7 @@ static void *qpa_thread_out (void *arg)
             }
         }
 
-        decr = to_mix = audio_MIN (pa->live, glob_paaudio.samples >> 2);
+        decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
         rpos = pa->rpos;
 
         if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {

@@ -314,7 +316,7 @@ static void *qpa_thread_in (void *arg)
             }
         }
 
-        incr = to_grab = audio_MIN (pa->dead, glob_paaudio.samples >> 2);
+        incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
         wpos = pa->wpos;
 
         if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {

@@ -430,7 +432,7 @@ static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
 
 static void context_state_cb (pa_context *c, void *userdata)
 {
-    paaudio *g = &glob_paaudio;
+    paaudio *g = userdata;
 
     switch (pa_context_get_state(c)) {
     case PA_CONTEXT_READY:

@@ -449,7 +451,7 @@ static void context_state_cb (pa_context *c, void *userdata)
 
 static void stream_state_cb (pa_stream *s, void * userdata)
 {
-    paaudio *g = &glob_paaudio;
+    paaudio *g = userdata;
 
     switch (pa_stream_get_state (s)) {
 

@@ -467,23 +469,21 @@ 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 = &glob_paaudio;
+    paaudio *g = userdata;
 
     pa_threaded_mainloop_signal (g->mainloop, 0);
 }
 
 static pa_stream *qpa_simple_new (
-        const char *server,
+        paaudio *g,
         const char *name,
         pa_stream_direction_t dir,
         const char *dev,
-        const char *stream_name,
         const pa_sample_spec *ss,
         const pa_channel_map *map,
         const pa_buffer_attr *attr,
         int *rerror)
 {
-    paaudio *g = &glob_paaudio;
     int r;
     pa_stream *stream;
 

@@ -538,12 +538,11 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
                         void *drv_opaque)
 {
     int error;
-    static pa_sample_spec ss;
-    static pa_buffer_attr ba;
+    pa_sample_spec ss;
+    pa_buffer_attr ba;
     struct audsettings obt_as = *as;
     PAVoiceOut *pa = (PAVoiceOut *) hw;
-
-    (void) drv_opaque;
+    paaudio *g = pa->g = drv_opaque;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;

@@ -561,11 +560,10 @@ 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 (
-        glob_paaudio.server,
+        g,
         "qemu",
         PA_STREAM_PLAYBACK,
-        glob_paaudio.sink,
-        "pcm.playback",
+        g->conf.sink,
         &ss,
         NULL,                   /* channel map */
         &ba,                    /* buffering attributes */

@@ -577,7 +575,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = glob_paaudio.samples;
+    hw->samples = g->conf.samples;
     pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
     pa->rpos = hw->rpos;
     if (!pa->pcm_buf) {

@@ -607,11 +605,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
 static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 {
     int error;
-    static pa_sample_spec ss;
+    pa_sample_spec ss;
     struct audsettings obt_as = *as;
     PAVoiceIn *pa = (PAVoiceIn *) hw;
-
-    (void) drv_opaque;
+    paaudio *g = pa->g = drv_opaque;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;

@@ -620,11 +617,10 @@ 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 (
-        glob_paaudio.server,
+        g,
         "qemu",
         PA_STREAM_RECORD,
-        glob_paaudio.source,
-        "pcm.capture",
+        g->conf.source,
         &ss,
         NULL,                   /* channel map */
         NULL,                   /* buffering attributes */

@@ -636,7 +632,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = glob_paaudio.samples;
+    hw->samples = g->conf.samples;
     pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
     pa->wpos = hw->wpos;
     if (!pa->pcm_buf) {

@@ -708,7 +704,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
     PAVoiceOut *pa = (PAVoiceOut *) hw;
     pa_operation *op;
     pa_cvolume v;
-    paaudio *g = &glob_paaudio;
+    paaudio *g = pa->g;
 
 #ifdef PA_CHECK_VERSION    /* macro is present in 0.9.16+ */
     pa_cvolume_init (&v);  /* function is present in 0.9.13+ */

@@ -760,7 +756,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
     PAVoiceIn *pa = (PAVoiceIn *) hw;
     pa_operation *op;
     pa_cvolume v;
-    paaudio *g = &glob_paaudio;
+    paaudio *g = pa->g;
 
 #ifdef PA_CHECK_VERSION
     pa_cvolume_init (&v);

@@ -810,23 +806,29 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 }
 
 /* common */
+static PAConf glob_conf = {
+    .samples = 4096,
+};
+
 static void *qpa_audio_init (void)
 {
-    paaudio *g = &glob_paaudio;
+    paaudio *g = g_malloc(sizeof(paaudio));
+    g->conf = glob_conf;
 
     g->mainloop = pa_threaded_mainloop_new ();
     if (!g->mainloop) {
         goto fail;
     }
 
-    g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop), glob_paaudio.server);
+    g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
+                                 g->conf.server);
     if (!g->context) {
         goto fail;
     }
 
     pa_context_set_state_callback (g->context, context_state_cb, g);
 
-    if (pa_context_connect (g->context, glob_paaudio.server, 0, NULL) < 0) {
+    if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
         qpa_logerr (pa_context_errno (g->context),
                     "pa_context_connect() failed\n");
         goto fail;

@@ -859,12 +861,13 @@ static void *qpa_audio_init (void)
 
     pa_threaded_mainloop_unlock (g->mainloop);
 
-    return &glob_paaudio;
+    return g;
 
 unlock_and_fail:
     pa_threaded_mainloop_unlock (g->mainloop);
 fail:
     AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+    g_free(g);
     return NULL;
 }
 

@@ -879,39 +882,38 @@ static void qpa_audio_fini (void *opaque)
     if (g->context) {
         pa_context_disconnect (g->context);
         pa_context_unref (g->context);
-        g->context = NULL;
     }
 
     if (g->mainloop) {
         pa_threaded_mainloop_free (g->mainloop);
     }
 
-    g->mainloop = NULL;
+    g_free(g);
 }
 
 struct audio_option qpa_options[] = {
     {
         .name  = "SAMPLES",
         .tag   = AUD_OPT_INT,
-        .valp  = &glob_paaudio.samples,
+        .valp  = &glob_conf.samples,
         .descr = "buffer size in samples"
     },
     {
         .name  = "SERVER",
         .tag   = AUD_OPT_STR,
-        .valp  = &glob_paaudio.server,
+        .valp  = &glob_conf.server,
         .descr = "server address"
     },
     {
         .name  = "SINK",
         .tag   = AUD_OPT_STR,
-        .valp  = &glob_paaudio.sink,
+        .valp  = &glob_conf.sink,
         .descr = "sink device name"
     },
     {
         .name  = "SOURCE",
         .tag   = AUD_OPT_STR,
-        .valp  = &glob_paaudio.source,
+        .valp  = &glob_conf.source,
         .descr = "source device name"
     },
     { /* End of list */ }

index 80ac610..d247969 100644
@@ -38,6 +38,17 @@
 #define USE_DSP_POLICY
 #endif
 
+typedef struct OSSConf {
+    int try_mmap;
+    int nfrags;
+    int fragsize;
+    const char *devpath_out;
+    const char *devpath_in;
+    int debug;
+    int exclusive;
+    int policy;
+} OSSConf;
+
 typedef struct OSSVoiceOut {
     HWVoiceOut hw;
     void *pcm_buf;

@@ -47,6 +58,7 @@ typedef struct OSSVoiceOut {
     int fragsize;
     int mmapped;
     int pending;
+    OSSConf *conf;
 } OSSVoiceOut;
 
 typedef struct OSSVoiceIn {

@@ -55,28 +67,9 @@ typedef struct OSSVoiceIn {
     int fd;
     int nfrags;
     int fragsize;
+    OSSConf *conf;
 } OSSVoiceIn;
 
-static struct {
-    int try_mmap;
-    int nfrags;
-    int fragsize;
-    const char *devpath_out;
-    const char *devpath_in;
-    int debug;
-    int exclusive;
-    int policy;
-} conf = {
-    .try_mmap = 0,
-    .nfrags = 4,
-    .fragsize = 4096,
-    .devpath_out = "/dev/dsp",
-    .devpath_in = "/dev/dsp",
-    .debug = 0,
-    .exclusive = 0,
-    .policy = 5
-};
-
 struct oss_params {
     int freq;
     audfmt_e fmt;

@@ -272,18 +265,18 @@ static int oss_get_version (int fd, int *version, const char *typ)
 #endif
 
 static int oss_open (int in, struct oss_params *req,
-                     struct oss_params *obt, int *pfd)
+                     struct oss_params *obt, int *pfd, OSSConf* conf)
 {
     int fd;
-    int oflags = conf.exclusive ? O_EXCL : 0;
+    int oflags = conf->exclusive ? O_EXCL : 0;
     audio_buf_info abinfo;
     int fmt, freq, nchannels;
     int setfragment = 1;
-    const char *dspname = in ? conf.devpath_in : conf.devpath_out;
+    const char *dspname = in ? conf->devpath_in : conf->devpath_out;
     const char *typ = in ? "ADC" : "DAC";
 
     /* Kludge needed to have working mmap on Linux */
-    oflags |= conf.try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+    oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
 
     fd = open (dspname, oflags | O_NONBLOCK);
     if (-1 == fd) {

@@ -317,20 +310,20 @@ static int oss_open (int in, struct oss_params *req,
     }
 
 #ifdef USE_DSP_POLICY
-    if (conf.policy >= 0) {
+    if (conf->policy >= 0) {
         int version;
 
         if (!oss_get_version (fd, &version, typ)) {
-            if (conf.debug) {
+            if (conf->debug) {
                 dolog ("OSS version = %#x\n", version);
             }
 
             if (version >= 0x040000) {
-                int policy = conf.policy;
+                int policy = conf->policy;
                 if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
                     oss_logerr2 (errno, typ,
                                  "Failed to set timing policy to %d\n",
-                                 conf.policy);
+                                 conf->policy);
                     goto err;
                 }
                 setfragment = 0;

@@ -434,6 +427,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
     struct audio_buf_info abinfo;
     struct count_info cntinfo;
     int bufsize;
+    OSSConf *conf = oss->conf;
 
     bufsize = hw->samples << hw->info.shift;
 

@@ -458,7 +452,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
         }
 
         if (abinfo.bytes > bufsize) {
-            if (conf.debug) {
+            if (conf->debug) {
                 dolog ("warning: Invalid available size, size=%d bufsize=%d\n"
                        "please report your OS/audio hw to [email protected]\n",
                        abinfo.bytes, bufsize);

@@ -467,7 +461,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
         }
 
         if (abinfo.bytes < 0) {
-            if (conf.debug) {
+            if (conf->debug) {
                 dolog ("warning: Invalid available size, size=%d bufsize=%d\n",
                        abinfo.bytes, bufsize);
             }

@@ -520,18 +514,17 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     int fd;
     audfmt_e effective_fmt;
     struct audsettings obt_as;
-
-    (void) drv_opaque;
+    OSSConf *conf = drv_opaque;
 
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.fragsize = conf.fragsize;
-    req.nfrags = conf.nfrags;
+    req.fragsize = conf->fragsize;
+    req.nfrags = conf->nfrags;
 
-    if (oss_open (0, &req, &obt, &fd)) {
+    if (oss_open (0, &req, &obt, &fd, conf)) {
         return -1;
     }
 

@@ -558,7 +551,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
 
     oss->mmapped = 0;
-    if (conf.try_mmap) {
+    if (conf->try_mmap) {
         oss->pcm_buf = mmap (
             NULL,
             hw->samples << hw->info.shift,

@@ -618,6 +611,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     oss->fd = fd;
+    oss->conf = conf;
     return 0;
 }
 

@@ -688,17 +682,16 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     int fd;
     audfmt_e effective_fmt;
     struct audsettings obt_as;
-
-    (void) drv_opaque;
+    OSSConf *conf = drv_opaque;
 
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.fragsize = conf.fragsize;
-    req.nfrags = conf.nfrags;
-    if (oss_open (1, &req, &obt, &fd)) {
+    req.fragsize = conf->fragsize;
+    req.nfrags = conf->nfrags;
+    if (oss_open (1, &req, &obt, &fd, conf)) {
         return -1;
     }
 

@@ -732,6 +725,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     oss->fd = fd;
+    oss->conf = conf;
     return 0;
 }
 

@@ -850,69 +844,83 @@ static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
     return 0;
 }
 
+static OSSConf glob_conf = {
+    .try_mmap = 0,
+    .nfrags = 4,
+    .fragsize = 4096,
+    .devpath_out = "/dev/dsp",
+    .devpath_in = "/dev/dsp",
+    .debug = 0,
+    .exclusive = 0,
+    .policy = 5
+};
+
 static void *oss_audio_init (void)
 {
-    if (access(conf.devpath_in, R_OK | W_OK) < 0 ||
-        access(conf.devpath_out, R_OK | W_OK) < 0) {
+    OSSConf *conf = g_malloc(sizeof(OSSConf));
+    *conf = glob_conf;
+
+    if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
+        access(conf->devpath_out, R_OK | W_OK) < 0) {
         return NULL;
     }
-    return &conf;
+    return conf;
 }
 
 static void oss_audio_fini (void *opaque)
 {
-    (void) opaque;
+    g_free(opaque);
 }
 
 static struct audio_option oss_options[] = {
     {
         .name  = "FRAGSIZE",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.fragsize,
+        .valp  = &glob_conf.fragsize,
         .descr = "Fragment size in bytes"
     },
     {
         .name  = "NFRAGS",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.nfrags,
+        .valp  = &glob_conf.nfrags,
         .descr = "Number of fragments"
     },
     {
         .name  = "MMAP",
         .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.try_mmap,
+        .valp  = &glob_conf.try_mmap,
         .descr = "Try using memory mapped access"
     },
     {
         .name  = "DAC_DEV",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.devpath_out,
+        .valp  = &glob_conf.devpath_out,
         .descr = "Path to DAC device"
     },
     {
         .name  = "ADC_DEV",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.devpath_in,
+        .valp  = &glob_conf.devpath_in,
         .descr = "Path to ADC device"
     },
     {
         .name  = "EXCLUSIVE",
         .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.exclusive,
+        .valp  = &glob_conf.exclusive,
         .descr = "Open device in exclusive mode (vmix wont work)"
     },
 #ifdef USE_DSP_POLICY
     {
         .name  = "POLICY",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.policy,
+        .valp  = &glob_conf.policy,
         .descr = "Set the timing policy of the device, -1 to use fragment mode",
     },
 #endif
     {
         .name  = "DEBUG",
         .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.debug,
+        .valp  = &glob_conf.debug,
         .descr = "Turn on some debugging messages"
     },
     { /* End of list */ }

index 5132aed..8acea8f 100644
@@ -36,15 +36,10 @@ typedef struct WAVVoiceOut {
     int total_samples;
 } WAVVoiceOut;
 
-static struct {
+typedef struct {
     struct audsettings settings;
     const char *wav_path;
-} conf = {
-    .settings.freq      = 44100,
-    .settings.nchannels = 2,
-    .settings.fmt       = AUD_FMT_S16,
-    .wav_path           = "qemu.wav"
-};
+} WAVConf;
 
 static int wav_run_out (HWVoiceOut *hw, int live)
 {

@@ -116,7 +111,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
         0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
         0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
     };
-    struct audsettings wav_as = conf.settings;
+    WAVConf *conf = drv_opaque;
+    struct audsettings wav_as = conf->settings;
 
     (void) as;
     (void) drv_opaque;

@@ -157,10 +153,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
     le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
     le_store (hdr + 32, 1 << (bits16 + stereo), 2);
 
-    wav->f = fopen (conf.wav_path, "wb");
+    wav->f = fopen (conf->wav_path, "wb");
     if (!wav->f) {
         dolog ("Failed to open wave file `%s'\nReason: %s\n",
-               conf.wav_path, strerror (errno));
+               conf->wav_path, strerror (errno));
         g_free (wav->pcm_buf);
         wav->pcm_buf = NULL;
         return -1;

@@ -228,40 +224,49 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
+static WAVConf glob_conf = {
+    .settings.freq      = 44100,
+    .settings.nchannels = 2,
+    .settings.fmt       = AUD_FMT_S16,
+    .wav_path           = "qemu.wav"
+};
+
 static void *wav_audio_init (void)
 {
-    return &conf;
+    WAVConf *conf = g_malloc(sizeof(WAVConf));
+    *conf = glob_conf;
+    return conf;
 }
 
 static void wav_audio_fini (void *opaque)
 {
-    (void) opaque;
     ldebug ("wav_fini");
+    g_free(opaque);
 }
 
 static struct audio_option wav_options[] = {
     {
         .name  = "FREQUENCY",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.settings.freq,
+        .valp  = &glob_conf.settings.freq,
         .descr = "Frequency"
     },
     {
         .name  = "FORMAT",
         .tag   = AUD_OPT_FMT,
-        .valp  = &conf.settings.fmt,
+        .valp  = &glob_conf.settings.fmt,
         .descr = "Format"
     },
     {
         .name  = "DAC_FIXED_CHANNELS",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.settings.nchannels,
+        .valp  = &glob_conf.settings.nchannels,
         .descr = "Number of channels (1 - mono, 2 - stereo)"
     },
     {
         .name  = "PATH",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.wav_path,
+        .valp  = &glob_conf.wav_path,
         .descr = "Path to wave file"
     },
     { /* End of list */ }

index 98276fb..85ba858 100644
@@ -67,7 +67,8 @@ static int glue (dsound_lock_, TYPE) (
     LPVOID *p2p,
     DWORD *blen1p,
     DWORD *blen2p,
-    int entire
+    int entire,
+    dsound *s
     )
 {
     HRESULT hr;

@@ -75,13 +76,14 @@ static int glue (dsound_lock_, TYPE) (
     LPVOID p1 = NULL, p2 = NULL;
     DWORD blen1 = 0, blen2 = 0;
     DWORD flag;
+    DSoundConf *conf = &s->conf;
 
 #ifdef DSBTYPE_IN
     flag = entire ? DSCBLOCK_ENTIREBUFFER : 0;
 #else
     flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
 #endif
-    for (i = 0; i < conf.lock_retries; ++i) {
+    for (i = 0; i < conf->lock_retries; ++i) {
         hr = glue (IFACE, _Lock) (
             buf,
             pos,

@@ -96,7 +98,7 @@ static int glue (dsound_lock_, TYPE) (
         if (FAILED (hr)) {
 #ifndef DSBTYPE_IN
             if (hr == DSERR_BUFFERLOST) {
-                if (glue (dsound_restore_, TYPE) (buf)) {
+                if (glue (dsound_restore_, TYPE) (buf, s)) {
                     dsound_logerr (hr, "Could not lock " NAME "\n");
                     goto fail;
                 }

@@ -110,7 +112,7 @@ static int glue (dsound_lock_, TYPE) (
         break;
     }
 
-    if (i == conf.lock_retries) {
+    if (i == conf->lock_retries) {
         dolog ("%d attempts to lock " NAME " failed\n", i);
         goto fail;
     }

@@ -183,9 +185,10 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
 {
     int err;
     HRESULT hr;
-    dsound *s = &glob_dsound;
+    dsound *s = drv_opaque;
     WAVEFORMATEX wfx;
     struct audsettings obt_as;
+    DSoundConf *conf = &s->conf;
 #ifdef DSBTYPE_IN
     const char *typ = "ADC";
     DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;

@@ -212,7 +215,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
     bd.dwSize = sizeof (bd);
     bd.lpwfxFormat = &wfx;
 #ifdef DSBTYPE_IN
-    bd.dwBufferBytes = conf.bufsize_in;
+    bd.dwBufferBytes = conf->bufsize_in;
     hr = IDirectSoundCapture_CreateCaptureBuffer (
         s->dsound_capture,
         &bd,

@@ -221,7 +224,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
         );
 #else
     bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
-    bd.dwBufferBytes = conf.bufsize_out;
+    bd.dwBufferBytes = conf->bufsize_out;
     hr = IDirectSound_CreateSoundBuffer (
         s->dsound,
         &bd,

@@ -271,6 +274,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
             );
     }
     hw->samples = bc.dwBufferBytes >> hw->info.shift;
+    ds->s = s;
 
 #ifdef DEBUG_DSOUND
     dolog ("caps %ld, desc %ld\n",

347 lines not on this page · raw message

index 35e8887..fea6071 100644
@@ -47,6 +47,8 @@ typedef struct {
     paaudio *g;
 } PAVoiceIn;
 
+static void qpa_audio_fini(void *opaque);
+
 static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
 {
     va_list ap;

@@ -814,6 +816,8 @@ static void *qpa_audio_init (void)
 {
     paaudio *g = g_malloc(sizeof(paaudio));
     g->conf = glob_conf;
+    g->mainloop = NULL;
+    g->context = NULL;
 
     g->mainloop = pa_threaded_mainloop_new ();
     if (!g->mainloop) {

@@ -867,7 +871,7 @@ unlock_and_fail:
     pa_threaded_mainloop_unlock (g->mainloop);
 fail:
     AUD_log (AUDIO_CAP, "Failed to initialize PA context");
-    g_free(g);
+    qpa_audio_fini(g);
     return NULL;
 }
 

index f38fd82..6dfd63e 100644
@@ -32,20 +32,16 @@
 #define AUDIO_CAP "coreaudio"
 #include "audio_int.h"
 
-struct {
+static int isAtexit;
+
+typedef struct {
     int buffer_frames;
     int nbuffers;
-    int isAtexit;
-} conf = {
-    .buffer_frames = 512,
-    .nbuffers = 4,
-    .isAtexit = 0
-};
+} CoreaudioConf;
 
 typedef struct coreaudioVoiceOut {
     HWVoiceOut hw;
     pthread_mutex_t mutex;
-    int isAtexit;
     AudioDeviceID outputDeviceID;
     UInt32 audioDevicePropertyBufferFrameSize;
     AudioStreamBasicDescription outputStreamBasicDescription;

@@ -161,7 +157,7 @@ static inline UInt32 isPlaying (AudioDeviceID outputDeviceID)
 
 static void coreaudio_atexit (void)
 {
-    conf.isAtexit = 1;
+    isAtexit = 1;
 }
 
 static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)

@@ -296,8 +292,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
     int err;
     const char *typ = "playback";
     AudioValueRange frameRange;
-
-    (void) drv_opaque;
+    CoreaudioConf *conf = drv_opaque;
 
     /* create mutex */
     err = pthread_mutex_init(&core->mutex, NULL);

@@ -339,16 +334,16 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
         return -1;
     }
 
-    if (frameRange.mMinimum > conf.buffer_frames) {
+    if (frameRange.mMinimum > conf->buffer_frames) {
         core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
         dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
     }
-    else if (frameRange.mMaximum < conf.buffer_frames) {
+    else if (frameRange.mMaximum < conf->buffer_frames) {
         core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
         dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
     }
     else {
-        core->audioDevicePropertyBufferFrameSize = conf.buffer_frames;
+        core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
     }
 
     /* set Buffer Frame Size */

@@ -382,7 +377,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
                            "Could not get device buffer frame size\n");
         return -1;
     }
-    hw->samples = conf.nbuffers * core->audioDevicePropertyBufferFrameSize;
+    hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
 
     /* get StreamFormat */
     propertySize = sizeof(core->outputStreamBasicDescription);

@@ -446,7 +441,7 @@ static void coreaudio_fini_out (HWVoiceOut *hw)
     int err;
     coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
 
-    if (!conf.isAtexit) {
+    if (!isAtexit) {
         /* stop playback */
         if (isPlaying(core->outputDeviceID)) {
             status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);

@@ -489,7 +484,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
 
     case VOICE_DISABLE:
         /* stop playback */
-        if (!conf.isAtexit) {
+        if (!isAtexit) {
             if (isPlaying(core->outputDeviceID)) {
                 status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
                 if (status != kAudioHardwareNoError) {

@@ -502,28 +497,36 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
+static CoreaudioConf glob_conf = {
+    .buffer_frames = 512,
+    .nbuffers = 4,
+};
+
 static void *coreaudio_audio_init (void)
 {
+    CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
+    *conf = glob_conf;
+
     atexit(coreaudio_atexit);
-    return &coreaudio_audio_init;
+    return conf;
 }
 
 static void coreaudio_audio_fini (void *opaque)
 {
-    (void) opaque;
+    g_free(opaque);
 }
 
 static struct audio_option coreaudio_options[] = {
     {
         .name  = "BUFFER_SIZE",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.buffer_frames,
+        .valp  = &glob_conf.buffer_frames,
         .descr = "Size of the buffer in frames"
     },
     {
         .name  = "BUFFER_COUNT",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.nbuffers,
+        .valp  = &glob_conf.nbuffers,
         .descr = "Number of buffers"
     },
     { /* End of list */ }