[Spice-devel] [spice-gtk PATCH v5 3/4] audio: spice-gstaudio implements async volume-info
Marc-André Lureau
marcandre.lureau at gmail.com
Fri Apr 3 07:25:57 PDT 2015
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.
+
> + 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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