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[Qemu-devel] [4827] Pulseaudio driver


From: malc
Subject: [Qemu-devel] [4827] Pulseaudio driver
Date: Wed, 02 Jul 2008 21:03:09 +0000

Revision: 4827
          http://svn.sv.gnu.org/viewvc/?view=rev&root=qemu&revision=4827
Author:   malc
Date:     2008-07-02 21:03:08 +0000 (Wed, 02 Jul 2008)

Log Message:
-----------
Pulseaudio driver

Modified Paths:
--------------
    trunk/Makefile
    trunk/Makefile.target
    trunk/audio/audio_int.h
    trunk/configure

Added Paths:
-----------
    trunk/audio/paaudio.c

Modified: trunk/Makefile
===================================================================
--- trunk/Makefile      2008-07-02 18:13:46 UTC (rev 4826)
+++ trunk/Makefile      2008-07-02 21:03:08 UTC (rev 4827)
@@ -98,6 +98,11 @@
 AUDIO_PT_INT = yes
 AUDIO_OBJS += esdaudio.o
 endif
+ifdef CONFIG_PA
+AUDIO_PT = yes
+AUDIO_PT_INT = yes
+AUDIO_OBJS += paaudio.o
+endif
 ifdef AUDIO_PT
 LDFLAGS += -pthread
 endif

Modified: trunk/Makefile.target
===================================================================
--- trunk/Makefile.target       2008-07-02 18:13:46 UTC (rev 4826)
+++ trunk/Makefile.target       2008-07-02 21:03:08 UTC (rev 4827)
@@ -270,6 +270,7 @@
        $(AR) rcs $@ $(LIBOBJS)
 
 translate.o: translate.c cpu.h $(OPC_H)
+translate.o: CFLAGS:=${CFLAGS} -O1 #-fno-unit-at-a-time
 
 translate-all.o: translate-all.c cpu.h $(OPC_H)
 
@@ -480,6 +481,9 @@
 ifdef CONFIG_ESD
 LIBS += -lesd
 endif
+ifdef CONFIG_PA
+LIBS += -lpulse-simple
+endif
 ifdef CONFIG_DSOUND
 LIBS += -lole32 -ldxguid
 endif

Modified: trunk/audio/audio_int.h
===================================================================
--- trunk/audio/audio_int.h     2008-07-02 18:13:46 UTC (rev 4826)
+++ trunk/audio/audio_int.h     2008-07-02 21:03:08 UTC (rev 4827)
@@ -203,6 +203,7 @@
 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 volume_t nominal_volume;
 
 void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);

Added: trunk/audio/paaudio.c
===================================================================
--- trunk/audio/paaudio.c                               (rev 0)
+++ trunk/audio/paaudio.c       2008-07-02 21:03:08 UTC (rev 4827)
@@ -0,0 +1,515 @@
+/* public domain */
+#include "qemu-common.h"
+#include "audio.h"
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+
+#define AUDIO_CAP "pulseaudio"
+#include "audio_int.h"
+#include "audio_pt_int.h"
+
+typedef struct {
+    HWVoiceOut hw;
+    int done;
+    int live;
+    int decr;
+    int rpos;
+    pa_simple *s;
+    void *pcm_buf;
+    struct audio_pt pt;
+} PAVoiceOut;
+
+typedef struct {
+    HWVoiceIn hw;
+    int done;
+    int dead;
+    int incr;
+    int wpos;
+    pa_simple *s;
+    void *pcm_buf;
+    struct audio_pt pt;
+} PAVoiceIn;
+
+static struct {
+    int samples;
+    int divisor;
+    char *server;
+    char *sink;
+    char *source;
+} conf = {
+    1024,
+    2,
+    NULL,
+    NULL,
+    NULL
+};
+
+static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
+{
+    va_list ap;
+
+    va_start (ap, fmt);
+    AUD_vlog (AUDIO_CAP, fmt, ap);
+    va_end (ap);
+
+    AUD_log (AUDIO_CAP, "Reason: %s\n", pa_strerror (err));
+}
+
+static void *qpa_thread_out (void *arg)
+{
+    PAVoiceOut *pa = arg;
+    HWVoiceOut *hw = &pa->hw;
+    int threshold;
+
+    threshold = conf.divisor ? hw->samples / conf.divisor : 0;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return NULL;
+    }
+
+    for (;;) {
+        int decr, to_mix, rpos;
+
+        for (;;) {
+            if (pa->done) {
+                goto exit;
+            }
+
+            if (pa->live > threshold) {
+                break;
+            }
+
+            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
+                goto exit;
+            }
+        }
+
+        decr = to_mix = pa->live;
+        rpos = hw->rpos;
+
+        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        while (to_mix) {
+            int error;
+            int chunk = audio_MIN (to_mix, hw->samples - rpos);
+            st_sample_t *src = hw->mix_buf + rpos;
+
+            hw->clip (pa->pcm_buf, src, chunk);
+
+            if (pa_simple_write (pa->s, pa->pcm_buf,
+                                 chunk << hw->info.shift, &error) < 0) {
+                qpa_logerr (error, "pa_simple_write failed\n");
+                return NULL;
+            }
+
+            rpos = (rpos + chunk) % hw->samples;
+            to_mix -= chunk;
+        }
+
+        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        pa->rpos = rpos;
+        pa->live -= decr;
+        pa->decr += decr;
+    }
+
+ exit:
+    audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    return NULL;
+}
+
+static int qpa_run_out (HWVoiceOut *hw)
+{
+    int live, decr;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_out (hw);
+    decr = audio_MIN (live, pa->decr);
+    pa->decr -= decr;
+    pa->live = live - decr;
+    hw->rpos = pa->rpos;
+    if (pa->live > 0) {
+        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    }
+    else {
+        audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    }
+    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)
+{
+    PAVoiceIn *pa = arg;
+    HWVoiceIn *hw = &pa->hw;
+    int threshold;
+
+    threshold = conf.divisor ? hw->samples / conf.divisor : 0;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return NULL;
+    }
+
+    for (;;) {
+        int incr, to_grab, wpos;
+
+        for (;;) {
+            if (pa->done) {
+                goto exit;
+            }
+
+            if (pa->dead > threshold) {
+                break;
+            }
+
+            if (audio_pt_wait (&pa->pt, AUDIO_FUNC)) {
+                goto exit;
+            }
+        }
+
+        incr = to_grab = pa->dead;
+        wpos = hw->wpos;
+
+        if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        while (to_grab) {
+            int error;
+            int chunk = audio_MIN (to_grab, hw->samples - wpos);
+            void *buf = advance (pa->pcm_buf, wpos);
+
+            if (pa_simple_read (pa->s, buf,
+                                chunk << hw->info.shift, &error) < 0) {
+                qpa_logerr (error, "pa_simple_read failed\n");
+                return NULL;
+            }
+
+            hw->conv (hw->conv_buf + wpos, buf, chunk, &nominal_volume);
+            wpos = (wpos + chunk) % hw->samples;
+            to_grab -= chunk;
+        }
+
+        if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+            return NULL;
+        }
+
+        pa->wpos = wpos;
+        pa->dead -= incr;
+        pa->incr += incr;
+    }
+
+ exit:
+    audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    return NULL;
+}
+
+static int qpa_run_in (HWVoiceIn *hw)
+{
+    int live, incr, dead;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return 0;
+    }
+
+    live = audio_pcm_hw_get_live_in (hw);
+    dead = hw->samples - live;
+    incr = audio_MIN (dead, pa->incr);
+    pa->incr -= incr;
+    pa->dead = dead - incr;
+    hw->wpos = pa->wpos;
+    if (pa->dead > 0) {
+        audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    }
+    else {
+        audio_pt_unlock (&pa->pt, AUDIO_FUNC);
+    }
+    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 (audfmt_e afmt, int endianness)
+{
+    int format;
+
+    switch (afmt) {
+    case AUD_FMT_S8:
+    case AUD_FMT_U8:
+        format = PA_SAMPLE_U8;
+        break;
+    case AUD_FMT_S16:
+    case AUD_FMT_U16:
+        format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
+        break;
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
+        break;
+    default:
+        dolog ("Internal logic error: Bad audio format %d\n", afmt);
+        format = PA_SAMPLE_U8;
+        break;
+    }
+    return format;
+}
+
+static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+{
+    switch (fmt) {
+    case PA_SAMPLE_U8:
+        return AUD_FMT_U8;
+    case PA_SAMPLE_S16BE:
+        *endianness = 1;
+        return AUD_FMT_S16;
+    case PA_SAMPLE_S16LE:
+        *endianness = 0;
+        return AUD_FMT_S16;
+    case PA_SAMPLE_S32BE:
+        *endianness = 1;
+        return AUD_FMT_S32;
+    case PA_SAMPLE_S32LE:
+        *endianness = 0;
+        return AUD_FMT_S32;
+    default:
+        dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
+        return AUD_FMT_U8;
+    }
+}
+
+static int qpa_init_out (HWVoiceOut *hw, audsettings_t *as)
+{
+    int error;
+    static pa_sample_spec ss;
+    audsettings_t obt_as = *as;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    ss.format = audfmt_to_pa (as->fmt, as->endianness);
+    ss.channels = as->nchannels;
+    ss.rate = as->freq;
+
+    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+
+    pa->s = pa_simple_new (
+        conf.server,
+        "qemu",
+        PA_STREAM_PLAYBACK,
+        conf.sink,
+        "pcm.playback",
+        &ss,
+        NULL,                   /* channel map */
+        NULL,                   /* buffering attributes */
+        &error
+        );
+    if (!pa->s) {
+        qpa_logerr (error, "pa_simple_new for playback failed\n");
+        goto fail1;
+    }
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = conf.samples;
+    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    if (!pa->pcm_buf) {
+        dolog ("Could not allocate buffer (%d bytes)\n",
+               hw->samples << hw->info.shift);
+        goto fail2;
+    }
+
+    if (audio_pt_init (&pa->pt, qpa_thread_out, hw, AUDIO_CAP, AUDIO_FUNC)) {
+        goto fail3;
+    }
+
+    return 0;
+
+ fail3:
+    free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    pa_simple_free (pa->s);
+    pa->s = NULL;
+ fail1:
+    return -1;
+}
+
+static int qpa_init_in (HWVoiceIn *hw, audsettings_t *as)
+{
+    int error;
+    static pa_sample_spec ss;
+    audsettings_t obt_as = *as;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+
+    ss.format = audfmt_to_pa (as->fmt, as->endianness);
+    ss.channels = as->nchannels;
+    ss.rate = as->freq;
+
+    obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+
+    pa->s = pa_simple_new (
+        conf.server,
+        "qemu",
+        PA_STREAM_RECORD,
+        conf.source,
+        "pcm.capture",
+        &ss,
+        NULL,                   /* channel map */
+        NULL,                   /* buffering attributes */
+        &error
+        );
+    if (!pa->s) {
+        qpa_logerr (error, "pa_simple_new for capture failed\n");
+        goto fail1;
+    }
+
+    audio_pcm_init_info (&hw->info, &obt_as);
+    hw->samples = conf.samples;
+    pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+    if (!pa->pcm_buf) {
+        dolog ("Could not allocate buffer (%d bytes)\n",
+               hw->samples << hw->info.shift);
+        goto fail2;
+    }
+
+    if (audio_pt_init (&pa->pt, qpa_thread_in, hw, AUDIO_CAP, AUDIO_FUNC)) {
+        goto fail3;
+    }
+
+    return 0;
+
+ fail3:
+    free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    pa_simple_free (pa->s);
+    pa->s = NULL;
+ fail1:
+    return -1;
+}
+
+static void qpa_fini_out (HWVoiceOut *hw)
+{
+    void *ret;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    audio_pt_lock (&pa->pt, AUDIO_FUNC);
+    pa->done = 1;
+    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
+
+    if (pa->s) {
+        pa_simple_free (pa->s);
+        pa->s = NULL;
+    }
+
+    audio_pt_fini (&pa->pt, AUDIO_FUNC);
+    qemu_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+}
+
+static void qpa_fini_in (HWVoiceIn *hw)
+{
+    void *ret;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+
+    audio_pt_lock (&pa->pt, AUDIO_FUNC);
+    pa->done = 1;
+    audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
+    audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
+
+    if (pa->s) {
+        pa_simple_free (pa->s);
+        pa->s = NULL;
+    }
+
+    audio_pt_fini (&pa->pt, AUDIO_FUNC);
+    qemu_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+}
+
+static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+    (void) hw;
+    (void) cmd;
+    return 0;
+}
+
+/* common */
+static void *qpa_audio_init (void)
+{
+    return &conf;
+}
+
+static void qpa_audio_fini (void *opaque)
+{
+    (void) opaque;
+}
+
+struct audio_option qpa_options[] = {
+    {"SAMPLES", AUD_OPT_INT, &conf.samples,
+     "buffer size in samples", NULL, 0},
+
+    {"DIVISOR", AUD_OPT_INT, &conf.divisor,
+     "threshold divisor", NULL, 0},
+
+    {"SERVER", AUD_OPT_STR, &conf.server,
+     "server address", NULL, 0},
+
+    {"SINK", AUD_OPT_STR, &conf.sink,
+     "sink device name", NULL, 0},
+
+    {"SOURCE", AUD_OPT_STR, &conf.source,
+     "source device name", NULL, 0},
+
+    {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+struct audio_pcm_ops qpa_pcm_ops = {
+    qpa_init_out,
+    qpa_fini_out,
+    qpa_run_out,
+    qpa_write,
+    qpa_ctl_out,
+    qpa_init_in,
+    qpa_fini_in,
+    qpa_run_in,
+    qpa_read,
+    qpa_ctl_in
+};
+
+struct audio_driver pa_audio_driver = {
+    INIT_FIELD (name           = ) "pa",
+    INIT_FIELD (descr          = ) "http://www.pulseaudio.org/";,
+    INIT_FIELD (options        = ) qpa_options,
+    INIT_FIELD (init           = ) qpa_audio_init,
+    INIT_FIELD (fini           = ) qpa_audio_fini,
+    INIT_FIELD (pcm_ops        = ) &qpa_pcm_ops,
+    INIT_FIELD (can_be_default = ) 0,
+    INIT_FIELD (max_voices_out = ) INT_MAX,
+    INIT_FIELD (max_voices_in  = ) INT_MAX,
+    INIT_FIELD (voice_size_out = ) sizeof (PAVoiceOut),
+    INIT_FIELD (voice_size_in  = ) sizeof (PAVoiceIn)
+};

Modified: trunk/configure
===================================================================
--- trunk/configure     2008-07-02 18:13:46 UTC (rev 4826)
+++ trunk/configure     2008-07-02 21:03:08 UTC (rev 4827)
@@ -196,7 +196,7 @@
 ;;
 *)
 audio_drv_list="oss"
-audio_possible_drivers="oss alsa sdl esd"
+audio_possible_drivers="oss alsa sdl esd pa"
 linux="yes"
 linux_user="yes"
 if [ "$cpu" = "i386" -o "$cpu" = "x86_64" ] ; then
@@ -767,6 +767,12 @@
     esd)
     audio_drv_probe $drv esd.h -lesd 'return esd_play_stream(0, 0, "", 0);'
     ;;
+
+    pa)
+    audio_drv_probe $drv pulse/simple.h -lpulse-simple \
+        "pa_simple *s = NULL; pa_simple_free(s); return 0;"
+    ;;
+
     esac
 done
 






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