static gboolean
gst_inter_audio_sink_stop (GstBaseSink * sink)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG_OBJECT (interaudiosink, "stop");
g_mutex_lock (&interaudiosink->surface->mutex);
gst_adapter_clear (interaudiosink->surface->audio_adapter);
memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
g_mutex_unlock (&interaudiosink->surface->mutex);
gst_inter_surface_unref (interaudiosink->surface);
interaudiosink->surface = NULL;
gst_adapter_clear (interaudiosink->input_adapter);
return TRUE;
}
static gboolean
gst_inter_audio_sink_start (GstBaseSink * sink)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG_OBJECT (interaudiosink, "start");
interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
g_mutex_lock (&interaudiosink->surface->mutex);
memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
/* We want to write latency-time before syncing has happened */
/* FIXME: The other side can change this value when it starts */
gst_base_sink_set_render_delay (sink,
interaudiosink->surface->audio_latency_time);
g_mutex_unlock (&interaudiosink->surface->mutex);
return TRUE;
}
static void
gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
if (interaudiosink->info.rate > 0) {
*end = *start +
gst_util_uint64_scale_int (gst_buffer_get_size (buffer), GST_SECOND,
interaudiosink->info.rate * interaudiosink->info.bpf);
}
}
}
}
static gboolean
gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (sink, "Failed to parse caps %" GST_PTR_FORMAT, caps);
return FALSE;
}
g_mutex_lock (&interaudiosink->surface->mutex);
interaudiosink->surface->audio_info = info;
interaudiosink->info = info;
/* TODO: Ideally we would drain the source here */
gst_adapter_clear (interaudiosink->surface->audio_adapter);
g_mutex_unlock (&interaudiosink->surface->mutex);
return TRUE;
}
static GstFlowReturn
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
int n;
GST_DEBUG ("render %d", GST_BUFFER_SIZE (buffer));
g_mutex_lock (interaudiosink->surface->mutex);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
if (n > (800 * 2 * 2)) {
GST_INFO ("flushing 800 samples");
gst_adapter_flush (interaudiosink->surface->audio_adapter, 800 * 4);
n -= 800;
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
g_mutex_unlock (interaudiosink->surface->mutex);
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:{
GstBuffer *tmp;
guint n;
if ((n = gst_adapter_available (interaudiosink->input_adapter)) > 0) {
g_mutex_lock (&interaudiosink->surface->mutex);
tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
g_mutex_unlock (&interaudiosink->surface->mutex);
}
break;
}
default:
break;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
