[Spice-devel] [spice-gtk PATCH v6 3/4] audio: spice-gstaudio implements async volume-info

Marc-André Lureau marcandre.lureau at gmail.com
Wed Apr 15 03:18:05 PDT 2015


looks good, ack

On Tue, Apr 14, 2015 at 2:18 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 by PulseAudio
> itself when requested)
> ---
>  gtk/spice-gstaudio.c | 191
> ++++++++++++++++++++++++++++++++++++++++++++++++++-
>  1 file changed, 190 insertions(+), 1 deletion(-)
>
> diff --git a/gtk/spice-gstaudio.c b/gtk/spice-gstaudio.c
> index 892028c..33de8e8 100644
> --- a/gtk/spice-gstaudio.c
> +++ b/gtk/spice-gstaudio.c
> @@ -50,6 +50,16 @@ 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,
> +        GCancellable *cancellable, SpiceMainChannel *main_channel,
> +        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,
> +        GCancellable *cancellable, SpiceMainChannel *main_channel,
> +        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 +118,10 @@ 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 +384,7 @@ 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 +410,7 @@ 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 +443,7 @@ 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 +472,7 @@ 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 +560,175 @@ SpiceGstaudio *spice_gstaudio_new(SpiceSession
> *session, GMainContext *context,
>
>      return gstaudio;
>  }
> +
> +static void spice_gstaudio_get_playback_volume_info_async(SpiceAudio
> *audio,
> +                                                          GCancellable
> *cancellable,
> +
> SpiceMainChannel *main_channel,
> +
> 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_check_cancellable (simple, cancellable);
> +
> +    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);
> +
> +    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);
> +    }
> +    g_object_unref(e);
> +
> +    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], 100*vol);
> +        }
> +    }
> +
> +    return g_simple_async_result_get_op_res_gboolean(simple);
> +}
> +
> +static void spice_gstaudio_get_record_volume_info_async(SpiceAudio *audio,
> +                                                        GCancellable
> *cancellable,
> +                                                        SpiceMainChannel
> *main_channel,
> +
> 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_check_cancellable (simple, cancellable);
> +
> +    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);
> +    }
> +    g_object_unref(e);
> +
> +    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], 100*vol);
> +        }
> +    }
> +
> +    return g_simple_async_result_get_op_res_gboolean(simple);
> +}
> --
> 2.1.0
>
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> Spice-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/spice-devel
>



-- 
Marc-André Lureau
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