FaSTTUBe/GST-Tensordecoder-ov_ep
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interaudiosink: Make sure to store at least period-time samples at once in the adapter

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It's better to create some more silence than having the source pull multiple
short buffers out of the adapter and insert silence between them.
This commit is contained in:
Sebastian Dröge 2014-11-12 18:06:45 +01:00
parent 5977ef5552
commit 6c3b3bef4e
2 changed files with 48 additions and 2 deletions
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49
gst/inter/gstinteraudiosink.c
View File

@ -63,6 +63,8 @@ static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
GstCaps * caps);
static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
GstEvent * event);
static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
static gboolean gst_inter_audio_sink_query (GstBaseSink * sink,
@ -83,6 +85,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
);
/* class initialization */
#define parent_class gst_inter_audio_sink_parent_class
G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
static void
@ -110,6 +113,7 @@ gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
base_sink_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_query);
@ -124,6 +128,7 @@ static void
gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink)
{
interaudiosink->channel = g_strdup ("default");
interaudiosink->input_adapter = gst_adapter_new ();
}
void
@ -166,6 +171,7 @@ gst_inter_audio_sink_finalize (GObject * object)
/* clean up object here */
g_free (interaudiosink->channel);
gst_object_unref (interaudiosink->input_adapter);
G_OBJECT_CLASS (gst_inter_audio_sink_parent_class)->finalize (object);
}
@ -225,6 +231,8 @@ gst_inter_audio_sink_stop (GstBaseSink * sink)
gst_inter_surface_unref (interaudiosink->surface);
interaudiosink->surface = NULL;
gst_adapter_clear (interaudiosink->input_adapter);
return TRUE;
}
@ -249,6 +257,31 @@ gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
return TRUE;
}
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);
}
static GstFlowReturn
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
@ -280,8 +313,20 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
period_samples * bpf);
n -= period_samples;
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
n = gst_adapter_available (interaudiosink->input_adapter);
if (period_samples * bpf > gst_buffer_get_size (buffer) + n) {
gst_adapter_push (interaudiosink->input_adapter, gst_buffer_ref (buffer));
} else {
GstBuffer *tmp;
if (n > 0) {
tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
}
g_mutex_unlock (&interaudiosink->surface->mutex);
return GST_FLOW_OK;

1
gst/inter/gstinteraudiosink.h
View File

@ -41,6 +41,7 @@ struct _GstInterAudioSink
GstInterSurface *surface;
char *channel;
GstAdapter *input_adapter;
GstAudioInfo info;
};