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rtpopuspay: support libwebrtc-compatible multichannel payload

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When the audio has more than 2 channels, add optional fields to output
caps from which webrtcbin can generate SDP in the syntax recognized by
"multiopus" codec present in libwebrtc [1].

e.g. for 5.1 audio:

a=rtpmap:96 multiopus/48000/6
a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5

[1] https://webrtc-review.googlesource.com/c/src/+/129768

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
This commit is contained in:
Jakub Adam 2020-11-30 21:49:48 +01:00 committed by GStreamer Merge Bot
parent 8b4147c757
commit b9ed5c7fa0
1 changed files with 56 additions and 8 deletions
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64
gst/rtp/gstrtpopuspay.c
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@ -36,10 +36,12 @@ GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0")
GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0;"
"audio/x-opus, channel-mapping-family = (int) 0, channels = (int) [1, 2];"
"audio/x-opus, channel-mapping-family = (int) 1, channels = (int) [3, 255]")
);
static GstStaticPadTemplate gst_rtp_opus_pay_src_template =
@ -50,8 +52,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 48000, "
"encoding-params = (string) \"2\", "
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"multiopus\" }")
);
static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
@ -103,7 +104,9 @@ gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
GstCaps *src_caps;
GstStructure *s, *outcaps;
const char *encoding_name = "OPUS";
gint channels, rate;
gint channels = 2;
gint rate;
gchar *encoding_params;
outcaps = gst_structure_new_empty ("unused");
@ -130,15 +133,60 @@ gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels)) {
if (channels > 2) {
GST_ERROR_OBJECT (payload,
"More than 2 channels with channel-mapping-family=0 is invalid");
return FALSE;
/* Implies channel-mapping-family = 1. */
gint stream_count, coupled_count;
const GValue *channel_mapping_array;
/* libwebrtc only supports "multiopus" when channels > 2. Mono and stereo
* sound must always be payloaded according to RFC 7587. */
encoding_name = "multiopus";
if (gst_structure_get_int (s, "stream-count", &stream_count)) {
char *num_streams = g_strdup_printf ("%d", stream_count);
gst_structure_set (outcaps, "num_streams", G_TYPE_STRING, num_streams,
NULL);
g_free (num_streams);
}
if (gst_structure_get_int (s, "coupled-count", &coupled_count)) {
char *coupled_streams = g_strdup_printf ("%d", coupled_count);
gst_structure_set (outcaps, "coupled_streams", G_TYPE_STRING,
coupled_streams, NULL);
g_free (coupled_streams);
}
channel_mapping_array = gst_structure_get_value (s, "channel-mapping");
if (GST_VALUE_HOLDS_ARRAY (channel_mapping_array)) {
GString *str = g_string_new (NULL);
guint i;
for (i = 0; i < gst_value_array_get_size (channel_mapping_array); ++i) {
if (i != 0) {
g_string_append_c (str, ',');
}
g_string_append_printf (str, "%d",
g_value_get_int (gst_value_array_get_value (channel_mapping_array,
i)));
}
gst_structure_set (outcaps, "channel_mapping", G_TYPE_STRING, str->str,
NULL);
g_string_free (str, TRUE);
}
} else {
gst_structure_set (outcaps, "sprop-stereo", G_TYPE_STRING,
(channels == 2) ? "1" : "0", NULL);
/* RFC 7587 requires the number of channels always be 2. */
channels = 2;
}
}
encoding_params = g_strdup_printf ("%d", channels);
gst_structure_set (outcaps, "encoding-params", G_TYPE_STRING,
encoding_params, NULL);
g_free (encoding_params);
if (gst_structure_get_int (s, "rate", &rate)) {
gchar *sprop_maxcapturerate = g_strdup_printf ("%d", rate);