Add a PulseAudio audio backend for Linux.

Change-Id: Ifaf876c41ab6c7275ba7d1dc8e12139f62840cd6

3 files changed, 626 insertions, 4 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 1ae42b6..0f5ee9e 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -47,10 +47,6 @@
 #  define I(...)  ((void)0)
 #endif
 
-
-#define  STRINGIFY_(x)  #x
-#define  STRINGIFY(x)   STRINGIFY_(x)
-
 #define  DYNLINK_FUNCTIONS   \
     DYNLINK_FUNC(size_t,snd_pcm_sw_params_sizeof,(void))    \
     DYNLINK_FUNC(int,snd_pcm_hw_params_current,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \
diff --git a/audio/audio.c b/audio/audio.c
index 1980450..aa4102b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -42,6 +42,9 @@
 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
 
 static struct audio_driver *drvtab[] = {
+#ifdef CONFIG_PULSEAUDIO
+    &pa_audio_driver,
+#endif
 #ifdef CONFIG_ESD
     &esd_audio_driver,
 #endif
diff --git a/audio/paaudio.c b/audio/paaudio.c
new file mode 100644
index 0000000..ed2c58c
--- /dev/null
+++ b/audio/paaudio.c
@@ -0,0 +1,623 @@
+/* public domain */
+#include "qemu-common.h"
+#include "audio.h"
+
+#include <pulse/simple.h>
+#include <pulse/error.h>
+#include <dlfcn.h>
+
+#define AUDIO_CAP "pulseaudio"
+#include "audio_int.h"
+#include "audio_pt_int.h"
+
+#define DEBUG 1
+
+#if DEBUG
+#  include "qemu_debug.h"
+#  include <stdio.h>
+#  define D(...)  VERBOSE_PRINT(audio,__VA_ARGS__)
+#  define D_ACTIVE  VERBOSE_CHECK(audio)
+#  define O(...)  VERBOSE_PRINT(audioout,__VA_ARGS__)
+#  define I(...)  VERBOSE_PRINT(audioin,__VA_ARGS__)
+#else
+#  define D(...)  ((void)0)
+#  define D_ACTIVE 0
+#  define O(...)  ((void)0)
+#  define I(...)  ((void)0)
+#endif
+
+#define  DYNLINK_FUNCTIONS   \
+    DYNLINK_FUNC(pa_simple*,pa_simple_new,(const char* server,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 *error)) \
+    DYNLINK_FUNC(void,pa_simple_free,(pa_simple* s))\
+    DYNLINK_FUNC(int,pa_simple_write,(pa_simple* s, const void* data, size_t bytes, int* error))\
+    DYNLINK_FUNC(int,pa_simple_read,(pa_simple* s,void* data, size_t bytes, int* error))\
+    DYNLINK_FUNC(const char*,pa_strerror,(int error))\
+
+#define DYNLINK_FUNCTIONS_INIT \
+    pa_dynlink_init
+
+static void* pa_lib;
+
+#include "dynlink.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 = {
+    .samples = 1024,
+    .divisor = 2,
+};
+
+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", FF(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);
+            struct st_sample *src = hw->mix_buf + rpos;
+
+            hw->clip (pa->pcm_buf, src, chunk);
+
+            if (FF(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 decr;
+    int live;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+
+    live = audio_pcm_hw_get_live_out (hw);
+    if (!live) {
+        return 0;
+    }
+
+    if (audio_pt_lock (&pa->pt, AUDIO_FUNC)) {
+        return 0;
+    }
+
+    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 (FF(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, struct audsettings *as)
+{
+    int error;
+    static pa_sample_spec ss;
+    struct audsettings 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 = FF(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:
+    qemu_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    FF(pa_simple_free) (pa->s);
+    pa->s = NULL;
+ fail1:
+    return -1;
+}
+
+static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
+{
+    int error;
+    static pa_sample_spec ss;
+    struct audsettings 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 = FF(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:
+    qemu_free (pa->pcm_buf);
+    pa->pcm_buf = NULL;
+ fail2:
+    FF(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) {
+        FF(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) {
+        FF(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)
+{
+    void*  result = NULL;
+
+    D("%s: entering", __FUNCTION__);
+    pa_lib = dlopen( "libpulse-simple.so", RTLD_NOW );
+    if (pa_lib == NULL)
+        pa_lib = dlopen( "libpulse-simple.so.0", RTLD_NOW );
+
+    if (pa_lib == NULL) {
+        D("could not find libpulse on this system\n");
+        goto Exit;
+    }
+
+    if (pa_dynlink_init(pa_lib) < 0)
+        goto Fail;
+
+    {
+        pa_sample_spec  ss;
+        int             error;
+        pa_simple*      simple;
+
+        ss.format   = PA_SAMPLE_U8;
+        ss.rate     = 44100;
+        ss.channels = 1;
+
+        /* try to open it for playback */
+        simple = FF(pa_simple_new) (
+            conf.server,
+            "qemu",
+            PA_STREAM_PLAYBACK,
+            conf.sink,
+            "pcm.playback",
+            &ss,
+            NULL,                   /* channel map */
+            NULL,                   /* buffering attributes */
+            &error
+            );
+
+        if (simple == NULL) {
+            D("%s: error opening open pulse audio library: %s",
+              __FUNCTION__, FF(pa_strerror)(error));
+            goto Fail;
+        }
+        FF(pa_simple_free)(simple);
+    }
+
+    result = &conf;
+    goto Exit;
+
+Fail:
+    D("%s: failed to open library\n", __FUNCTION__);
+    dlclose(pa_lib);
+
+Exit:
+    D("%s: exiting", __FUNCTION__);
+    return result;
+}
+
+static void qpa_audio_fini (void *opaque)
+{
+    if (pa_lib != NULL) {
+        dlclose(pa_lib);
+        pa_lib = NULL;
+    }
+    (void) opaque;
+    (void) opaque;
+}
+
+struct audio_option qpa_options[] = {
+    {
+        .name  = "SAMPLES",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.samples,
+        .descr = "buffer size in samples"
+    },
+    {
+        .name  = "DIVISOR",
+        .tag   = AUD_OPT_INT,
+        .valp  = &conf.divisor,
+        .descr = "threshold divisor"
+    },
+    {
+        .name  = "SERVER",
+        .tag   = AUD_OPT_STR,
+        .valp  = &conf.server,
+        .descr = "server address"
+    },
+    {
+        .name  = "SINK",
+        .tag   = AUD_OPT_STR,
+        .valp  = &conf.sink,
+        .descr = "sink device name"
+    },
+    {
+        .name  = "SOURCE",
+        .tag   = AUD_OPT_STR,
+        .valp  = &conf.source,
+        .descr = "source device name"
+    },
+    { /* End of list */ }
+};
+
+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
+};
+
+struct audio_driver pa_audio_driver = {
+    .name           = "pa",
+    .descr          = "http://www.pulseaudio.org/",
+    .options        = qpa_options,
+    .init           = qpa_audio_init,
+    .fini           = qpa_audio_fini,
+    .pcm_ops        = &qpa_pcm_ops,
+    .can_be_default = 1,
+    .max_voices_out = INT_MAX,
+    .max_voices_in  = INT_MAX,
+    .voice_size_out = sizeof (PAVoiceOut),
+    .voice_size_in  = sizeof (PAVoiceIn)
+};