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gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
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
parent
f1392c148e
commit
767ac56e4e
3 changed files with 51 additions and 25 deletions
16
ChangeLog
16
ChangeLog
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@ -1,3 +1,19 @@
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2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/audioresample/gstaudioresample.c:
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(gst_audioresample_class_init), (gst_audioresample_init),
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(audioresample_start), (audioresample_stop),
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(gst_audioresample_set_property), (gst_audioresample_get_property):
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Implement GstBaseTransform::start and ::stop so that audioresample
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can clear its internal state properly and be reused insted of
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causing non-negotiated errors with playbin under some circumstances
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(#342789).
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* tests/check/elements/audioresample.c: (setup_audioresample),
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(cleanup_audioresample):
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Need to set element state here so that ::start and ::stop are
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called.
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2006-06-16 Wim Taymans <wim@fluendo.com>
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Patch by: Young-Ho Cha <ganadist at chollian dot net>
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@ -60,13 +60,6 @@ GST_ELEMENT_DETAILS ("Audio scaler",
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"Resample audio",
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"David Schleef <ds@schleef.org>");
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/* GstAudioresample signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_FILTERLEN 16
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enum
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@ -111,8 +104,6 @@ static GstStaticPadTemplate gst_audioresample_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
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static void gst_audioresample_dispose (GObject * object);
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static void gst_audioresample_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audioresample_get_property (GObject * object,
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@ -133,8 +124,8 @@ static GstFlowReturn audioresample_pushthrough (GstAudioresample *
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static GstFlowReturn audioresample_transform (GstBaseTransform * base,
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GstBuffer * inbuf, GstBuffer * outbuf);
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static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
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/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
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static gboolean audioresample_start (GstBaseTransform * base);
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static gboolean audioresample_stop (GstBaseTransform * base);
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
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@ -164,13 +155,16 @@ gst_audioresample_class_init (GstAudioresampleClass * klass)
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gobject_class->set_property = gst_audioresample_set_property;
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gobject_class->get_property = gst_audioresample_get_property;
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gobject_class->dispose = gst_audioresample_dispose;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FILTERLEN,
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g_param_spec_int ("filter_length", "filter_length", "filter_length",
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0, G_MAXINT, DEFAULT_FILTERLEN,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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GST_BASE_TRANSFORM_CLASS (klass)->start =
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GST_DEBUG_FUNCPTR (audioresample_start);
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GST_BASE_TRANSFORM_CLASS (klass)->stop =
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GST_DEBUG_FUNCPTR (audioresample_stop);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
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GST_DEBUG_FUNCPTR (audioresample_transform_size);
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GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
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@ -191,7 +185,6 @@ static void
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gst_audioresample_init (GstAudioresample * audioresample,
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GstAudioresampleClass * klass)
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{
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ResampleState *r;
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GstBaseTransform *trans;
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trans = GST_BASE_TRANSFORM (audioresample);
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@ -200,29 +193,39 @@ gst_audioresample_init (GstAudioresample * audioresample,
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* is trivial in the passtrough case. */
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gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
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r = resample_new ();
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audioresample->resample = r;
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audioresample->filter_length = DEFAULT_FILTERLEN;
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}
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/* vmethods */
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static gboolean
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audioresample_start (GstBaseTransform * base)
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{
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GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
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audioresample->resample = resample_new ();
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audioresample->ts_offset = -1;
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audioresample->offset = -1;
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audioresample->next_ts = -1;
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resample_set_filter_length (r, DEFAULT_FILTERLEN);
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resample_set_filter_length (audioresample->resample,
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audioresample->filter_length);
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return TRUE;
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}
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static void
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gst_audioresample_dispose (GObject * object)
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static gboolean
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audioresample_stop (GstBaseTransform * base)
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{
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GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
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GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
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if (audioresample->resample) {
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resample_free (audioresample->resample);
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audioresample->resample = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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return TRUE;
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}
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/* vmethods */
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gboolean
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audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
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guint * size)
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@ -639,7 +642,6 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
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{
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GstAudioresample *audioresample;
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g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
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audioresample = GST_AUDIORESAMPLE (object);
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switch (prop_id) {
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@ -647,8 +649,10 @@ gst_audioresample_set_property (GObject * object, guint prop_id,
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audioresample->filter_length = g_value_get_int (value);
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GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
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audioresample->filter_length);
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if (audioresample->resample) {
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resample_set_filter_length (audioresample->resample,
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audioresample->filter_length);
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}
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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@ -662,7 +666,6 @@ gst_audioresample_get_property (GObject * object, guint prop_id,
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{
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GstAudioresample *audioresample;
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g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
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audioresample = GST_AUDIORESAMPLE (object);
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switch (prop_id) {
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@ -69,6 +69,10 @@ setup_audioresample (int channels, int inrate, int outrate)
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"rate", G_TYPE_INT, inrate, NULL);
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fail_unless (gst_caps_is_fixed (caps));
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
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"could not set to paused");
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mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
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pad = gst_pad_get_peer (mysrcpad);
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gst_pad_set_caps (pad, caps);
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@ -100,6 +104,9 @@ cleanup_audioresample (GstElement * audioresample)
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{
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GST_DEBUG ("cleanup_audioresample");
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fail_unless (gst_element_set_state (audioresample,
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GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
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gst_check_teardown_src_pad (audioresample);
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gst_check_teardown_sink_pad (audioresample);
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gst_check_teardown_element (audioresample);
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