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gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...

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Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
This commit is contained in:
Tim-Philipp Müller 2006-06-16 15:17:44 +00:00
parent f1392c148e
commit 767ac56e4e
3 changed files with 51 additions and 25 deletions
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16
ChangeLog
View file

@ -1,3 +1,19 @@
2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
2006-06-16 Wim Taymans <wim@fluendo.com>
Patch by: Young-Ho Cha <ganadist at chollian dot net>

53
gst/audioresample/gstaudioresample.c
View file

@ -60,13 +60,6 @@ GST_ELEMENT_DETAILS ("Audio scaler",
"Resample audio",
"David Schleef <ds@schleef.org>");
/* GstAudioresample signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_FILTERLEN 16
enum
@ -111,8 +104,6 @@ static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_dispose (GObject * object);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
@ -133,8 +124,8 @@ static GstFlowReturn audioresample_pushthrough (GstAudioresample *
static GstFlowReturn audioresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
static gboolean audioresample_start (GstBaseTransform * base);
static gboolean audioresample_stop (GstBaseTransform * base);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
@ -164,13 +155,16 @@ gst_audioresample_class_init (GstAudioresampleClass * klass)
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, DEFAULT_FILTERLEN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->start =
GST_DEBUG_FUNCPTR (audioresample_start);
GST_BASE_TRANSFORM_CLASS (klass)->stop =
GST_DEBUG_FUNCPTR (audioresample_stop);
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (audioresample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
@ -191,7 +185,6 @@ static void
gst_audioresample_init (GstAudioresample * audioresample,
GstAudioresampleClass * klass)
{
ResampleState *r;
GstBaseTransform *trans;
trans = GST_BASE_TRANSFORM (audioresample);
@ -200,29 +193,39 @@ gst_audioresample_init (GstAudioresample * audioresample,
* is trivial in the passtrough case. */
gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
r = resample_new ();
audioresample->resample = r;
audioresample->filter_length = DEFAULT_FILTERLEN;
}
/* vmethods */
static gboolean
audioresample_start (GstBaseTransform * base)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
audioresample->resample = resample_new ();
audioresample->ts_offset = -1;
audioresample->offset = -1;
audioresample->next_ts = -1;
resample_set_filter_length (r, DEFAULT_FILTERLEN);
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
return TRUE;
}
static void
gst_audioresample_dispose (GObject * object)
static gboolean
audioresample_stop (GstBaseTransform * base)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
if (audioresample->resample) {
resample_free (audioresample->resample);
audioresample->resample = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
return TRUE;
}
/* vmethods */
gboolean
audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size)
@ -639,7 +642,6 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
{
GstAudioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
@ -647,8 +649,10 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
if (audioresample->resample) {
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -662,7 +666,6 @@ gst_audioresample_get_property (GObject * object, guint prop_id,
{
GstAudioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {

7
tests/check/elements/audioresample.c
View file

@ -69,6 +69,10 @@ setup_audioresample (int channels, int inrate, int outrate)
"rate", G_TYPE_INT, inrate, NULL);
fail_unless (gst_caps_is_fixed (caps));
fail_unless (gst_element_set_state (audioresample,
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
"could not set to paused");
mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
pad = gst_pad_get_peer (mysrcpad);
gst_pad_set_caps (pad, caps);
@ -100,6 +104,9 @@ cleanup_audioresample (GstElement * audioresample)
{
GST_DEBUG ("cleanup_audioresample");
fail_unless (gst_element_set_state (audioresample,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
gst_check_teardown_src_pad (audioresample);
gst_check_teardown_sink_pad (audioresample);
gst_check_teardown_element (audioresample);