[Spice-devel] [spice-gtk PATCH v5 3/4] audio: spice-gstaudio implements async volume-info

Victor Toso victortoso at redhat.com
Fri Apr 3 07:53:50 PDT 2015


On Fri, Apr 03, 2015 at 04:25:57PM +0200, Marc-André Lureau wrote:
> On Fri, Apr 3, 2015 at 3:53 PM, Victor Toso <victortoso at redhat.com> wrote:
> 
> > Gstaudio rely on sink/src elements to get the volume/mute.
> > (e.g. pulsesink and pulsesrc, the values are updated)
> > ---
> >  gtk/spice-gstaudio.c | 179
> > ++++++++++++++++++++++++++++++++++++++++++++++++++-
> >  1 file changed, 178 insertions(+), 1 deletion(-)
> >
> > diff --git a/gtk/spice-gstaudio.c b/gtk/spice-gstaudio.c
> > index 892028c..845e5be 100644
> > --- a/gtk/spice-gstaudio.c
> > +++ b/gtk/spice-gstaudio.c
> > @@ -50,6 +50,14 @@ struct _SpiceGstaudioPrivate {
> >
> >  static gboolean connect_channel(SpiceAudio *audio, SpiceChannel *channel);
> >  static void channel_weak_notified(gpointer data, GObject
> > *where_the_object_was);
> > +static void spice_gstaudio_get_playback_volume_info_async(SpiceAudio
> > *audio,
> > +        GAsyncReadyCallback callback, gpointer user_data);
> > +static gboolean spice_gstaudio_get_playback_volume_info_finish(SpiceAudio
> > *audio,
> > +        GAsyncResult *res, gboolean *mute, guint8 *nchannels, guint16
> > **volume, GError **error);
> > +static void spice_gstaudio_get_record_volume_info_async(SpiceAudio *audio,
> > +        GAsyncReadyCallback callback, gpointer user_data);
> > +static gboolean spice_gstaudio_get_record_volume_info_finish(SpiceAudio
> > *audio,
> > +        GAsyncResult *res, gboolean *mute, guint8 *nchannels, guint16
> > **volume, GError **error);
> >
> >  static void spice_gstaudio_finalize(GObject *obj)
> >  {
> > @@ -108,6 +116,10 @@ static void
> > spice_gstaudio_class_init(SpiceGstaudioClass *klass)
> >      SpiceAudioClass *audio_class = SPICE_AUDIO_CLASS(klass);
> >
> >      audio_class->connect_channel = connect_channel;
> > +    audio_class->get_playback_volume_info_async =
> > spice_gstaudio_get_playback_volume_info_async;
> > +    audio_class->get_playback_volume_info_finish =
> > spice_gstaudio_get_playback_volume_info_finish;
> > +    audio_class->get_record_volume_info_async =
> > spice_gstaudio_get_record_volume_info_async;
> > +    audio_class->get_record_volume_info_finish =
> > spice_gstaudio_get_record_volume_info_finish;
> >
> >      gobject_class->finalize = spice_gstaudio_finalize;
> >      gobject_class->dispose = spice_gstaudio_dispose;
> > @@ -370,6 +382,7 @@ static void playback_volume_changed(GObject *object,
> > GParamSpec *pspec, gpointer
> >      g_return_if_fail(nchannels > 0);
> >
> >      vol = 1.0 * volume[0] / VOLUME_NORMAL;
> > +    SPICE_DEBUG("%s volume changed to %u (%0.2f)", __func__, volume[0],
> > 100*vol);
> >
> >      if (GST_IS_BIN(p->playback.sink))
> >          e = gst_bin_get_by_interface(GST_BIN(p->playback.sink),
> > GST_TYPE_STREAM_VOLUME);
> > @@ -395,7 +408,7 @@ static void playback_mute_changed(GObject *object,
> > GParamSpec *pspec, gpointer d
> >          return;
> >
> >      g_object_get(object, "mute", &mute, NULL);
> > -    SPICE_DEBUG("playback mute changed %u", mute);
> > +    SPICE_DEBUG("%s mute changed to %u", __func__, mute);
> >
> >      if (GST_IS_BIN(p->playback.sink))
> >          e = gst_bin_get_by_interface(GST_BIN(p->playback.sink),
> > GST_TYPE_STREAM_VOLUME);
> > @@ -428,6 +441,7 @@ static void record_volume_changed(GObject *object,
> > GParamSpec *pspec, gpointer d
> >      g_return_if_fail(nchannels > 0);
> >
> >      vol = 1.0 * volume[0] / VOLUME_NORMAL;
> > +    SPICE_DEBUG("%s volume changed to %u (%0.2f)", __func__, volume[0],
> > 100*vol);
> >
> >      /* TODO directsoundsrc doesn't support IDirectSoundBuffer_SetVolume */
> >      /* TODO pulsesrc doesn't support volume property, it's all coming! */
> > @@ -456,6 +470,7 @@ static void record_mute_changed(GObject *object,
> > GParamSpec *pspec, gpointer dat
> >          return;
> >
> >      g_object_get(object, "mute", &mute, NULL);
> > +    SPICE_DEBUG("%s mute changed to %u", __func__, mute);
> >
> >      if (GST_IS_BIN(p->record.src))
> >          e = gst_bin_get_by_interface(GST_BIN(p->record.src),
> > GST_TYPE_STREAM_VOLUME);
> > @@ -543,3 +558,165 @@ SpiceGstaudio *spice_gstaudio_new(SpiceSession
> > *session, GMainContext *context,
> >
> >      return gstaudio;
> >  }
> > +
> > +static void spice_gstaudio_get_playback_volume_info_async(SpiceAudio
> > *audio,
> > +
> > GAsyncReadyCallback callback,
> > +                                                          gpointer
> > user_data)
> > +{
> > +    GSimpleAsyncResult *simple;
> > +
> > +    simple = g_simple_async_result_new(G_OBJECT(audio),
> > +                                       callback,
> > +                                       user_data,
> > +
> >  spice_gstaudio_get_playback_volume_info_async);
> > +    g_simple_async_result_set_op_res_gboolean(simple, TRUE);
> > +    g_simple_async_result_complete_in_idle(simple);
> > +}
> > +
> > +static gboolean spice_gstaudio_get_playback_volume_info_finish(SpiceAudio
> > *audio,
> > +
> >  GAsyncResult *res,
> > +                                                               gboolean
> > *mute,
> > +                                                               guint8
> > *nchannels,
> > +                                                               guint16
> > **volume,
> > +                                                               GError
> > **error)
> > +{
> > +    SpiceGstaudioPrivate *p = SPICE_GSTAUDIO(audio)->priv;
> > +    GstElement *e;
> > +    gboolean lmute;
> > +    gdouble vol;
> > +    gboolean fake_channel = FALSE;
> > +    GSimpleAsyncResult *simple = (GSimpleAsyncResult *) res;
> > +
> > +    g_return_val_if_fail(g_simple_async_result_is_valid(res,
> > +        G_OBJECT(audio), spice_gstaudio_get_playback_volume_info_async),
> > FALSE);
> > +
> > +    if (g_simple_async_result_propagate_error(simple, error)) {
> > +        return FALSE;
> > +    }
> > +
> > +    if (p->playback.sink == NULL || p->playback.channels == 0) {
> > +        SPICE_DEBUG("%s PlaybackChannel not created yet, force start",
> > __func__);
> > +        /* In order to get system volume, we start the pipeline */
> > +        playback_start(NULL, SPICE_AUDIO_FMT_S16, 2, 48000, audio);
> > +        fake_channel = TRUE;
> > +    }
> > +
> > +    if (GST_IS_BIN(p->playback.sink))
> > +        e = gst_bin_get_by_interface(GST_BIN(p->playback.sink),
> > GST_TYPE_STREAM_VOLUME);
> > +    else
> > +        e = g_object_ref(p->playback.sink);
> >
> 
> I can't see where is corresponding the unref.

Because I forgot, many thanks.
Fixed.

> 
> +
> > +    if (GST_IS_STREAM_VOLUME(e)) {
> > +        vol = gst_stream_volume_get_volume(GST_STREAM_VOLUME(e),
> > GST_STREAM_VOLUME_FORMAT_CUBIC);
> > +        lmute = gst_stream_volume_get_mute(GST_STREAM_VOLUME(e));
> > +    } else {
> > +        g_object_get(e,
> > +                     "volume", &vol,
> > +                     "mute", &lmute, NULL);
> > +    }
> > +
> > +    if (fake_channel) {
> > +        SPICE_DEBUG("%s Stop faked PlaybackChannel", __func__);
> > +        playback_stop(NULL, audio);
> > +    }
> > +
> > +    if (mute != NULL) {
> > +        *mute = lmute;
> > +    }
> > +
> > +    if (nchannels != NULL) {
> > +        *nchannels = p->playback.channels;
> > +    }
> > +
> > +    if (volume != NULL) {
> > +        gint i;
> > +        *volume = g_new(guint16, p->playback.channels);
> > +        for (i = 0; i < p->playback.channels; i++) {
> > +            (*volume)[i] = (guint16) (vol * VOLUME_NORMAL);
> > +            SPICE_DEBUG("(playback) volume at %d is %u (%0.2f%%)", i,
> > (*volume)[i], vol);
> > +        }
> > +    }
> > +
> > +    return g_simple_async_result_get_op_res_gboolean(simple);
> > +}
> > +
> > +static void spice_gstaudio_get_record_volume_info_async(SpiceAudio *audio,
> > +
> > GAsyncReadyCallback callback,
> > +                                                        gpointer
> > user_data)
> > +{
> > +    GSimpleAsyncResult *simple;
> > +
> > +    simple = g_simple_async_result_new(G_OBJECT(audio),
> > +                                       callback,
> > +                                       user_data,
> > +
> >  spice_gstaudio_get_record_volume_info_async);
> > +    g_simple_async_result_set_op_res_gboolean(simple, TRUE);
> > +    g_simple_async_result_complete_in_idle(simple);
> > +}
> > +
> > +static gboolean spice_gstaudio_get_record_volume_info_finish(SpiceAudio
> > *audio,
> > +                                                             GAsyncResult
> > *res,
> > +                                                             gboolean
> > *mute,
> > +                                                             guint8
> > *nchannels,
> > +                                                             guint16
> > **volume,
> > +                                                             GError
> > **error)
> > +{
> > +    SpiceGstaudioPrivate *p = SPICE_GSTAUDIO(audio)->priv;
> > +    GstElement *e;
> > +    gboolean lmute;
> > +    gdouble vol;
> > +    gboolean fake_channel = FALSE;
> > +    GSimpleAsyncResult *simple = (GSimpleAsyncResult *) res;
> > +
> > +    g_return_val_if_fail(g_simple_async_result_is_valid(res,
> > +        G_OBJECT(audio), spice_gstaudio_get_record_volume_info_async),
> > FALSE);
> > +
> > +    if (g_simple_async_result_propagate_error(simple, error)) {
> > +        return FALSE;
> > +    }
> > +
> > +    if (p->record.src == NULL || p->record.channels == 0) {
> > +        SPICE_DEBUG("%s RecordChannel not created yet, force start",
> > __func__);
> > +        /* In order to get system volume, we start the pipeline */
> > +        record_start(NULL, SPICE_AUDIO_FMT_S16, 2, 48000, audio);
> > +        fake_channel = TRUE;
> > +    }
> > +
> > +    if (GST_IS_BIN(p->record.src))
> > +        e = gst_bin_get_by_interface(GST_BIN(p->record.src),
> > GST_TYPE_STREAM_VOLUME);
> > +    else
> > +        e = g_object_ref(p->record.src);
> > +
> > +    if (GST_IS_STREAM_VOLUME(e)) {
> > +        vol = gst_stream_volume_get_volume(GST_STREAM_VOLUME(e),
> > GST_STREAM_VOLUME_FORMAT_CUBIC);
> > +        lmute = gst_stream_volume_get_mute(GST_STREAM_VOLUME(e));
> > +    } else {
> > +        g_object_get(e,
> > +                     "volume", &vol,
> > +                     "mute", &lmute, NULL);
> > +    }
> > +
> > +    if (fake_channel) {
> > +        SPICE_DEBUG("%s Stop faked RecordChannel", __func__);
> > +        record_stop(SPICE_GSTAUDIO(audio));
> > +    }
> > +
> > +    if (mute != NULL) {
> > +        *mute = lmute;
> > +    }
> > +
> > +    if (nchannels != NULL) {
> > +        *nchannels = p->record.channels;
> > +    }
> > +
> > +    if (volume != NULL) {
> > +        gint i;
> > +        *volume = g_new(guint16, p->record.channels);
> > +        for (i = 0; i < p->record.channels; i++) {
> > +            (*volume)[i] = (guint16) (vol * VOLUME_NORMAL);
> > +            SPICE_DEBUG("(record) volume at %d is %u (%0.2f%%)", i,
> > (*volume)[i], vol);
> > +        }
> > +    }
> > +
> > +    return g_simple_async_result_get_op_res_gboolean(simple);
> > +}
> > --
> > 2.1.0
> >
> > _______________________________________________
> > Spice-devel mailing list
> > Spice-devel at lists.freedesktop.org
> > http://lists.freedesktop.org/mailman/listinfo/spice-devel
> >
> 
> 
> 
> -- 
> Marc-André Lureau


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