mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-26 11:41:09 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpopus: Put MULTIOPUS in all caps

Browse source 
The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
This commit is contained in:
Olivier Crête 2023-01-03 17:44:37 -05:00 committed by Olivier Crête
parent 085e6c036a
commit c51ae6112d
3 changed files with 8 additions and 8 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

4
subprojects/gst-plugins-good/docs/gst_plugins_cache.json
View file

@ -15572,7 +15572,7 @@
"long-name": "RTP Opus packet depayloader",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n payload: [ 96, 127 ]\n clock-rate: 48000\n encoding-name: { (string)OPUS, (string)X-GST-OPUS-DRAFT-SPITTKA-00, (string)multiopus }\n",
"caps": "application/x-rtp:\n media: audio\n payload: [ 96, 127 ]\n clock-rate: 48000\n encoding-name: { (string)OPUS, (string)X-GST-OPUS-DRAFT-SPITTKA-00, (string)MULTIOPUS }\n",
"direction": "sink",
"presence": "always"
},
@ -15605,7 +15605,7 @@
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: audio\n payload: [ 96, 127 ]\n clock-rate: 48000\n encoding-name: { (string)OPUS, (string)X-GST-OPUS-DRAFT-SPITTKA-00, (string)multiopus }\n",
"caps": "application/x-rtp:\n media: audio\n payload: [ 96, 127 ]\n clock-rate: 48000\n encoding-name: { (string)OPUS, (string)X-GST-OPUS-DRAFT-SPITTKA-00, (string)MULTIOPUS }\n",
"direction": "src",
"presence": "always"
}

6
subprojects/gst-plugins-good/gst/rtp/gstrtpopusdepay.c
View file

@ -42,7 +42,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
"clock-rate = (int) 48000, "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"multiopus\" }")
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"MULTIOPUS\" }")
);
static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
@ -105,7 +105,7 @@ gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
s = gst_caps_get_structure (caps, 0);
if (g_str_equal (gst_structure_get_string (s, "encoding-name"), "multiopus")) {
if (g_str_equal (gst_structure_get_string (s, "encoding-name"), "MULTIOPUS")) {
gint channels;
gint stream_count;
gint coupled_count;
@ -119,7 +119,7 @@ gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
!gst_structure_has_field_typed (s, "num_streams", G_TYPE_STRING) ||
!gst_structure_has_field_typed (s, "coupled_streams", G_TYPE_STRING) ||
!gst_structure_has_field_typed (s, "channel_mapping", G_TYPE_STRING)) {
GST_WARNING_OBJECT (depayload, "Encoding name 'multiopus' requires "
GST_WARNING_OBJECT (depayload, "Encoding name 'MULTIOPUS' requires "
"encoding-params, num_streams, coupled_streams and channel_mapping "
"as string fields in caps.");
goto reject_caps;

6
subprojects/gst-plugins-good/gst/rtp/gstrtpopuspay.c
View file

@ -28,7 +28,7 @@
*
* In addition to the RFC, which assumes only mono and stereo payload,
* the element supports multichannel Opus audio streams using a non-standardized
* SDP config and "multiopus" codec developed by Google for libwebrtc. When the
* SDP config and "MULTIOPUS" codec developed by Google for libwebrtc. When the
* input data have more than 2 channels, rtpopuspay will add extra fields to
* output caps that can be used to generate SDP in the syntax understood by
* libwebrtc. For example in the case of 5.1 audio:
@ -83,7 +83,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 48000, "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"multiopus\" }")
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"MULTIOPUS\" }")
);
static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
@ -255,7 +255,7 @@ gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
/* libwebrtc only supports "multiopus" when channels > 2. Mono and stereo
* sound must always be payloaded according to RFC 7587. */
encoding_name = "multiopus";
encoding_name = "MULTIOPUS";
if (gst_structure_get_int (s, "stream-count", &stream_count)) {
char *num_streams = g_strdup_printf ("%d", stream_count);