Commit 85aee8e2 authored by Andy Wingo's avatar Andy Wingo
Browse files

gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)

Original commit message from CVS:
2007-01-06  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
parent 4e3fc961
2007-01-06 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
 
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
......@@ -85,6 +85,8 @@ static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
basesink);
static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
gboolean active);
static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
......@@ -155,6 +157,8 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
gstbasesink_class->async_play =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
gstbasesink_class->activate_pull =
GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull);
}
static void
......@@ -167,6 +171,9 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
baseaudiosink->provided_clock = gst_audio_clock_new ("clock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
GST_BASE_SINK (baseaudiosink)->can_activate_pull = TRUE;
}
static void
......@@ -810,11 +817,80 @@ gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
return buffer;
}
static gboolean
gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
{
gboolean ret;
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
if (active) {
GstCaps *sinkcaps, *peercaps, *caps;
gst_ring_buffer_set_callback (sink->ringbuffer,
gst_base_audio_sink_callback, sink);
/* need to spawn a thread to start pulling. that's the ring buffer thread,
which is started in ring_buffer_acquire(), which is called due to a sink
setcaps(). So we need to setcaps, which is tough because we don't know
exactly what caps we'll be getting. We can guess, though, and that's as
good as we're going to get without the user telling us explicitly e.g.
via a capsfilter before the sink. */
sinkcaps = gst_pad_get_caps (basesink->sinkpad);
peercaps = gst_pad_peer_get_caps (basesink->sinkpad);
caps = gst_caps_intersect (sinkcaps, peercaps);
gst_caps_unref (sinkcaps);
gst_caps_unref (peercaps);
gst_caps_truncate (caps);
gst_pad_fixate_caps (basesink->sinkpad, caps);
GST_DEBUG_OBJECT (sink, "initializing pull-mode ringbuffer with caps %"
GST_PTR_FORMAT, caps);
ret = gst_pad_set_caps (basesink->sinkpad, caps);
gst_caps_unref (caps);
} else {
gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
/* stop thread */
ret = gst_ring_buffer_release (sink->ringbuffer);
}
return ret;
}
static void
gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
gpointer user_data)
{
/* GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (data); */
GstBaseSink *basesink;
GstBaseAudioSink *sink;
GstBuffer *buf;
GstFlowReturn ret;
basesink = GST_BASE_SINK (data);
sink = GST_BASE_AUDIO_SINK (data);
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
will copy twice, once into data, once into DMA */
GST_LOG_OBJECT (basesink, "pulling %d bytes to fill audio buffer", len);
ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf);
if (ret != GST_FLOW_OK)
goto error;
if (len != GST_BUFFER_SIZE (buf)) {
GST_INFO_OBJECT (basesink, "short read pulling from sink pad: %d<%d",
len, GST_BUFFER_SIZE (buf));
len = MIN (GST_BUFFER_SIZE (buf), len);
}
memcpy (data, GST_BUFFER_DATA (buf), len);
return;
error:
{
GST_WARNING_OBJECT (basesink, "Got flow error but can't return it: %d",
ret);
return;
}
}
/* should be called with the LOCK */
......@@ -884,8 +960,6 @@ gst_base_audio_sink_change_state (GstElement * element,
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL) {
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
gst_ring_buffer_set_callback (sink->ringbuffer,
gst_base_audio_sink_callback, sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment