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@xxxxxxxxxx> 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@xxxxxxxxxxxxxxxxxxxxx > > http://lists.freedesktop.org/mailman/listinfo/spice-devel > > > > > > -- > Marc-André Lureau _______________________________________________ Spice-devel mailing list Spice-devel@xxxxxxxxxxxxxxxxxxxxx http://lists.freedesktop.org/mailman/listinfo/spice-devel