From a5755233e874dc095a24cd9d2900336d9f693ffe Mon Sep 17 00:00:00 2001 From: David Schleef Date: Sun, 14 Sep 2003 11:20:45 +0000 Subject: [PATCH] Fixes to make it pass media test. Remove frequency parameter, since it can be (and should be) set by caps negotiation. Original commit message from CVS: Fixes to make it pass media test. Remove frequency parameter, since it can be (and should be) set by caps negotiation. --- gst/audioscale/gstaudioscale.c | 128 +++++++++++++++++++++++++++------ gst/audioscale/gstaudioscale.h | 1 + 2 files changed, 107 insertions(+), 22 deletions(-) diff --git a/gst/audioscale/gstaudioscale.c b/gst/audioscale/gstaudioscale.c index 46b83c58fc..36116a0665 100644 --- a/gst/audioscale/gstaudioscale.c +++ b/gst/audioscale/gstaudioscale.c @@ -48,7 +48,6 @@ enum { enum { ARG_0, - ARG_FREQUENCY, ARG_FILTERLEN, ARG_METHOD, /* FILL ME */ @@ -134,9 +133,6 @@ gst_audioscale_class_init (AudioscaleClass *klass) gobject_class = (GObjectClass*)klass; gstelement_class = (GstElementClass*)klass; - g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY, - g_param_spec_int ("frequency","frequency","frequency", - 0,G_MAXINT,44100,G_PARAM_READWRITE|G_PARAM_CONSTRUCT)); g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN, g_param_spec_int ("filter_length", "filter_length", "filter_length", 0, G_MAXINT, 16, G_PARAM_READWRITE|G_PARAM_CONSTRUCT)); @@ -151,31 +147,119 @@ gst_audioscale_class_init (AudioscaleClass *klass) } +static GstCaps * +gst_audioscale_getcaps (GstPad *pad, GstCaps *caps) +{ + Audioscale *audioscale; + GstCaps *peercaps; + + audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); + + if (pad == audioscale->srcpad){ + peercaps = gst_pad_get_allowed_caps (audioscale->sinkpad); + }else{ + peercaps = gst_pad_get_allowed_caps (audioscale->srcpad); + } + + if(peercaps == GST_CAPS_NONE){ + return GST_CAPS_NONE; + } + + caps = gst_caps_copy (peercaps); +#if 1 + /* we do this hack, because the audioscale lib doesn't handle + * rate conversions larger than a factor of 2 */ + if(gst_caps_has_property_typed(caps, "rate", GST_PROPS_INT_RANGE_TYPE)){ + int rate_min, rate_max; + + gst_props_entry_get_int_range (gst_props_get_entry(caps->properties, "rate"), + &rate_min, &rate_max); + gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE((rate_min+1)/2, + rate_max*2)); + }else{ + int rate; + + gst_caps_get_int (caps, "rate", &rate); + gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE((rate+1)/2,rate*2)); + } +#else + gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE(4000,96000)); +#endif + + return caps; +} + static GstPadLinkReturn -gst_audioscale_sinkconnect (GstPad * pad, GstCaps * caps) +gst_audioscale_sink_link (GstPad * pad, GstCaps * caps) { Audioscale *audioscale; resample_t *r; - GstCaps *newcaps; + GstCaps *caps1; + GstCaps *caps2; + GstCaps *peercaps; gint rate; + int ret; audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); r = audioscale->resample; + if (!GST_CAPS_IS_FIXED (caps)){ + return GST_PAD_LINK_DELAYED; + } + + ret = gst_pad_try_set_caps (audioscale->srcpad, caps); + + if(ret == GST_PAD_LINK_OK || ret == GST_PAD_LINK_DONE){ + audioscale->passthru = TRUE; + return ret; + } + + audioscale->passthru = FALSE; + gst_caps_get_int (caps, "rate", &rate); gst_caps_get_int (caps, "channels", &r->channels); r->i_rate = rate; - resample_reinit(r); - - newcaps = gst_caps_copy (caps); - gst_caps_set (newcaps, "rate", GST_PROPS_INT_TYPE, audioscale->targetfrequency, NULL); - if (GST_CAPS_IS_FIXED (caps)) - return gst_pad_try_set_caps (audioscale->srcpad, newcaps); - else - return GST_PAD_LINK_DELAYED; + peercaps = gst_pad_get_allowed_caps (audioscale->srcpad); + + caps1 = gst_caps_copy (caps); +#if 1 + /* we do this hack, because the audioscale lib doesn't handle + * rate conversions larger than a factor of 2 */ + if(gst_caps_has_property_typed(caps1, "rate", GST_PROPS_INT_RANGE_TYPE)){ + int rate_min, rate_max; + + gst_props_entry_get_int_range (gst_props_get_entry(caps1->properties, "rate"), + &rate_min, &rate_max); + gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE((rate_min+1)/2, + rate_max*2)); + }else{ + gst_caps_get_int (caps1, "rate", &rate); + gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE((rate+1)/2,rate*2)); + } +#else + gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE(4000,96000)); +#endif + caps2 = gst_caps_intersect(caps1, peercaps); + gst_caps_unref(caps1); + + if(caps2 == GST_CAPS_NONE){ + return GST_PAD_LINK_REFUSED; + } + + if (GST_CAPS_IS_FIXED (caps2)) { + ret = gst_pad_try_set_caps (audioscale->srcpad, caps2); + gst_caps_get_int (caps, "rate", &rate); + r->o_rate = rate; + audioscale->targetfrequency = rate; + resample_reinit(r); + return ret; + } + + gst_caps_unref (caps2); + return GST_PAD_LINK_DELAYED; } static void * @@ -201,12 +285,14 @@ gst_audioscale_init (Audioscale *audioscale) GST_PAD_TEMPLATE_GET (sink_factory), "sink"); gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->sinkpad); gst_pad_set_chain_function(audioscale->sinkpad,gst_audioscale_chain); - gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_sinkconnect); + gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_sink_link); + gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps); audioscale->srcpad = gst_pad_new_from_template ( GST_PAD_TEMPLATE_GET (src_factory), "src"); gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->srcpad); + gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps); r = g_new0(resample_t,1); audioscale->resample = r; @@ -238,6 +324,11 @@ gst_audioscale_chain (GstPad *pad, GstBuffer *buf) g_return_if_fail(buf != NULL); audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad)); + if (audioscale->passthru){ + gst_pad_push (audioscale->srcpad, buf); + return; + } + data = GST_BUFFER_DATA(buf); size = GST_BUFFER_SIZE(buf); @@ -266,10 +357,6 @@ gst_audioscale_set_property (GObject * object, guint prop_id, r = src->resample; switch (prop_id) { - case ARG_FREQUENCY: - src->targetfrequency = g_value_get_int (value); - r->o_rate = src->targetfrequency; - break; case ARG_FILTERLEN: r->filter_length = g_value_get_int (value); GST_DEBUG_OBJECT (GST_ELEMENT(src), "new filter length %d\n", r->filter_length); @@ -295,9 +382,6 @@ gst_audioscale_get_property (GObject *object, guint prop_id, GValue *value, GPar r = src->resample; switch (prop_id) { - case ARG_FREQUENCY: - g_value_set_int (value, src->targetfrequency); - break; case ARG_FILTERLEN: g_value_set_int (value, r->filter_length); break; diff --git a/gst/audioscale/gstaudioscale.h b/gst/audioscale/gstaudioscale.h index a8541e5334..ed930daac0 100644 --- a/gst/audioscale/gstaudioscale.h +++ b/gst/audioscale/gstaudioscale.h @@ -59,6 +59,7 @@ struct _Audioscale { GstPad *sinkpad,*srcpad; /* audio state */ + gboolean passthru; gint format; gint channels; gint frequency;