mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-24 18:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity

opusenc: add upstream negotiation for multistream ability

Browse source 
This will help elements that cannot deal with multistream,
such as the RTP payloader.

The caps now do not include a "streams" field anymore, but
a "multistream" boolean, since we have no real use for knowing
the exact amount of streams.

https://bugzilla.gnome.org/show_bug.cgi?id=665078
This commit is contained in:
Vincent Penquerc'h 2011-12-09 17:25:41 +00:00
parent 3a978f5a69
commit 3f47a43f2a
2 changed files with 69 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

66
ext/opus/gstopusenc.c
View file

@ -126,7 +126,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, streams = (int) [1, 255 ]")
GST_STATIC_CAPS ("audio/x-opus")
);
#define DEFAULT_AUDIO TRUE
@ -161,6 +161,7 @@ static void gst_opus_enc_finalize (GObject * object);
static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc,
GstEvent * event);
static GstCaps *gst_opus_enc_sink_getcaps (GstAudioEncoder * benc);
static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
static void gst_opus_enc_get_property (GObject * object, guint prop_id,
@ -229,6 +230,7 @@ gst_opus_enc_class_init (GstOpusEncClass * klass)
base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame);
base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps);
g_object_class_install_property (gobject_class, PROP_AUDIO,
g_param_spec_boolean ("audio", "Audio or voice",
@ -721,6 +723,68 @@ gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
return FALSE;
}
static GstCaps *
gst_opus_enc_sink_getcaps (GstAudioEncoder * benc)
{
GstOpusEnc *enc;
GstCaps *caps;
GstCaps *peercaps = NULL;
GstCaps *intersect = NULL;
guint i;
gboolean allow_multistream;
enc = GST_OPUS_ENC (benc);
GST_DEBUG_OBJECT (enc, "sink getcaps");
peercaps = gst_pad_peer_get_caps (GST_AUDIO_ENCODER_SRC_PAD (benc));
if (!peercaps) {
GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
return
gst_caps_copy (gst_pad_get_pad_template_caps
(GST_AUDIO_ENCODER_SINK_PAD (benc)));
}
intersect = gst_caps_intersect (peercaps,
gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (benc)));
gst_caps_unref (peercaps);
if (gst_caps_is_empty (intersect))
return intersect;
allow_multistream = FALSE;
for (i = 0; i < gst_caps_get_size (intersect); i++) {
GstStructure *s = gst_caps_get_structure (intersect, i);
gboolean multistream;
if (gst_structure_get_boolean (s, "multistream", &multistream)) {
if (multistream) {
allow_multistream = TRUE;
}
} else {
allow_multistream = TRUE;
}
}
gst_caps_unref (intersect);
caps =
gst_caps_copy (gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SINK_PAD
(benc)));
if (!allow_multistream) {
GValue range = { 0 };
g_value_init (&range, GST_TYPE_INT_RANGE);
gst_value_set_int_range (&range, 1, 2);
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *s = gst_caps_get_structure (caps, i);
gst_structure_set_value (s, "channels", &range);
}
g_value_unset (&range);
}
GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
return caps;
}
static GstFlowReturn
gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
{

7
ext/opus/gstopusheader.c
View file

@ -151,6 +151,7 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
GstBuffer * buf1, GstBuffer * buf2)
{
int n_streams, family;
gboolean multistream;
g_return_if_fail (caps);
g_return_if_fail (headers && !*headers);
@ -167,9 +168,9 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
}
/* mark and put on caps */
*caps =
gst_caps_new_simple ("audio/x-opus", "streams", G_TYPE_INT, n_streams,
NULL);
multistream = n_streams > 1;
*caps = gst_caps_new_simple ("audio/x-opus",
"multistream", G_TYPE_BOOLEAN, multistream, NULL);
*caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
*headers = g_slist_prepend (*headers, buf2);