opusenc: add upstream negotiation for multistream ability

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

diff --git a/ext/opus/gstopusenc.c b/ext/opus/gstopusenc.c
index 502f4a9..0ccbbeb 100644 (file)
--- a/ext/opus/gstopusenc.c
+++ b/ext/opus/gstopusenc.c
@@ -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)
 {

diff --git a/ext/opus/gstopusheader.c b/ext/opus/gstopusheader.c
index 185c85e..d3d631f 100644 (file)
--- a/ext/opus/gstopusheader.c
+++ b/ext/opus/gstopusheader.c
@@ -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);

diff --git a/ext/opus/gstopusparse.c b/ext/opus/gstopusparse.c
index 542031f..f3706cb 100644 (file)
--- a/ext/opus/gstopusparse.c
+++ b/ext/opus/gstopusparse.c
@@ -58,7 +58,7 @@ static GstStaticPadTemplate opus_parse_sink_factory =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-opus, streams = (int) [ 1, 255 ]")
+    GST_STATIC_CAPS ("audio/x-opus")
     );
 
 G_DEFINE_TYPE (GstOpusParse, gst_opus_parse, GST_TYPE_BASE_PARSE);

diff --git a/ext/opus/gstrtpopuspay.c b/ext/opus/gstrtpopuspay.c
index 9724e62..cdd7ee0 100644 (file)
--- a/ext/opus/gstrtpopuspay.c
+++ b/ext/opus/gstrtpopuspay.c
@@ -37,7 +37,7 @@ static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-opus")
+    GST_STATIC_CAPS ("audio/x-opus, multistream = (boolean) FALSE")
     );
 
 static GstStaticPadTemplate gst_rtp_opus_pay_src_template =