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vorbisenc: use more accurate sink pad template caps

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Removes the need for custom caps query handling and makes it more
correct from the beginning on the template. It is a bit uglier
to read because there is 1 entry per channel but makes code easier
to maintain.
This commit is contained in:
Thiago Santos 2015-08-16 12:20:51 -03:00
parent cd7c7dba70
commit 6a26a42ffd
1 changed files with 9 additions and 36 deletions
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45
ext/vorbis/gstvorbisenc.c
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@ -56,16 +56,6 @@
GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug);
#define GST_CAT_DEFAULT vorbisenc_debug
static GstStaticPadTemplate vorbis_enc_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, 200000 ], " "channels = (int) [ 1, 255 ]")
);
static GstStaticPadTemplate vorbis_enc_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
@ -86,6 +76,7 @@ enum
};
static GstFlowReturn gst_vorbis_enc_output_buffers (GstVorbisEnc * vorbisenc);
static GstCaps *gst_vorbis_enc_generate_sink_caps (void);
#define MAX_BITRATE_DEFAULT -1
@ -101,8 +92,6 @@ static gboolean gst_vorbis_enc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_vorbis_enc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstCaps *gst_vorbis_enc_getcaps (GstAudioEncoder * enc,
GstCaps * filter);
static gboolean gst_vorbis_enc_sink_event (GstAudioEncoder * enc,
GstEvent * event);
@ -125,6 +114,8 @@ gst_vorbis_enc_class_init (GstVorbisEncClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstAudioEncoderClass *base_class;
GstCaps *sink_caps;
GstPadTemplate *sink_templ;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
@ -165,10 +156,14 @@ gst_vorbis_enc_class_init (GstVorbisEncClass * klass)
"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
sink_caps = gst_vorbis_enc_generate_sink_caps ();
sink_templ = gst_pad_template_new ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (gstelement_class, sink_templ);
gst_caps_unref (sink_caps);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&vorbis_enc_src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&vorbis_enc_sink_factory));
gst_element_class_set_static_metadata (gstelement_class,
"Vorbis audio encoder", "Codec/Encoder/Audio",
@ -179,7 +174,6 @@ gst_vorbis_enc_class_init (GstVorbisEncClass * klass)
base_class->stop = GST_DEBUG_FUNCPTR (gst_vorbis_enc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_vorbis_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_vorbis_enc_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_vorbis_enc_getcaps);
base_class->sink_event = GST_DEBUG_FUNCPTR (gst_vorbis_enc_sink_event);
base_class->flush = GST_DEBUG_FUNCPTR (gst_vorbis_enc_flush);
}
@ -289,27 +283,6 @@ gst_vorbis_enc_generate_sink_caps (void)
return caps;
}
static GstCaps *
gst_vorbis_enc_getcaps (GstAudioEncoder * enc, GstCaps * filter)
{
GstVorbisEnc *vorbisenc = GST_VORBISENC (enc);
GstCaps *caps;
if (vorbisenc->sinkcaps == NULL)
vorbisenc->sinkcaps = gst_vorbis_enc_generate_sink_caps ();
if (filter) {
GstCaps *int_caps = gst_caps_intersect_full (filter, vorbisenc->sinkcaps,
GST_CAPS_INTERSECT_FIRST);
caps = gst_audio_encoder_proxy_getcaps (enc, int_caps, filter);
gst_caps_unref (int_caps);
} else {
caps = gst_audio_encoder_proxy_getcaps (enc, vorbisenc->sinkcaps, filter);
}
return caps;
}
static gint64
gst_vorbis_enc_get_latency (GstVorbisEnc * vorbisenc)
{