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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
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3a978f5a69
commit
3f47a43f2a
2 changed files with 69 additions and 4 deletions
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@ -126,7 +126,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-opus, streams = (int) [1, 255 ]")
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GST_STATIC_CAPS ("audio/x-opus")
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);
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#define DEFAULT_AUDIO TRUE
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@ -161,6 +161,7 @@ static void gst_opus_enc_finalize (GObject * object);
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static gboolean gst_opus_enc_sink_event (GstAudioEncoder * benc,
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GstEvent * event);
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static GstCaps *gst_opus_enc_sink_getcaps (GstAudioEncoder * benc);
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static gboolean gst_opus_enc_setup (GstOpusEnc * enc);
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static void gst_opus_enc_get_property (GObject * object, guint prop_id,
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@ -229,6 +230,7 @@ gst_opus_enc_class_init (GstOpusEncClass * klass)
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_opus_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_opus_enc_handle_frame);
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base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
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base_class->getcaps = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_getcaps);
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g_object_class_install_property (gobject_class, PROP_AUDIO,
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g_param_spec_boolean ("audio", "Audio or voice",
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@ -721,6 +723,68 @@ gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
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return FALSE;
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}
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static GstCaps *
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gst_opus_enc_sink_getcaps (GstAudioEncoder * benc)
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{
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GstOpusEnc *enc;
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GstCaps *caps;
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GstCaps *peercaps = NULL;
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GstCaps *intersect = NULL;
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guint i;
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gboolean allow_multistream;
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enc = GST_OPUS_ENC (benc);
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GST_DEBUG_OBJECT (enc, "sink getcaps");
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peercaps = gst_pad_peer_get_caps (GST_AUDIO_ENCODER_SRC_PAD (benc));
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if (!peercaps) {
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GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
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return
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gst_caps_copy (gst_pad_get_pad_template_caps
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(GST_AUDIO_ENCODER_SINK_PAD (benc)));
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}
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intersect = gst_caps_intersect (peercaps,
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gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (benc)));
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gst_caps_unref (peercaps);
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if (gst_caps_is_empty (intersect))
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return intersect;
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allow_multistream = FALSE;
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for (i = 0; i < gst_caps_get_size (intersect); i++) {
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GstStructure *s = gst_caps_get_structure (intersect, i);
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gboolean multistream;
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if (gst_structure_get_boolean (s, "multistream", &multistream)) {
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if (multistream) {
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allow_multistream = TRUE;
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}
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} else {
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allow_multistream = TRUE;
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}
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}
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gst_caps_unref (intersect);
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caps =
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gst_caps_copy (gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SINK_PAD
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(benc)));
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if (!allow_multistream) {
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GValue range = { 0 };
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g_value_init (&range, GST_TYPE_INT_RANGE);
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gst_value_set_int_range (&range, 1, 2);
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for (i = 0; i < gst_caps_get_size (caps); i++) {
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GstStructure *s = gst_caps_get_structure (caps, i);
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gst_structure_set_value (s, "channels", &range);
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}
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g_value_unset (&range);
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}
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GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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static GstFlowReturn
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gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
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{
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@ -151,6 +151,7 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
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GstBuffer * buf1, GstBuffer * buf2)
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{
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int n_streams, family;
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gboolean multistream;
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g_return_if_fail (caps);
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g_return_if_fail (headers && !*headers);
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@ -167,9 +168,9 @@ gst_opus_header_create_caps_from_headers (GstCaps ** caps, GSList ** headers,
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}
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/* mark and put on caps */
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*caps =
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gst_caps_new_simple ("audio/x-opus", "streams", G_TYPE_INT, n_streams,
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NULL);
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multistream = n_streams > 1;
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*caps = gst_caps_new_simple ("audio/x-opus",
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"multistream", G_TYPE_BOOLEAN, multistream, NULL);
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*caps = _gst_caps_set_buffer_array (*caps, "streamheader", buf1, buf2, NULL);
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*headers = g_slist_prepend (*headers, buf2);
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