gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpopuspay.c
Olivier Crête c272d0bfcd rtopuspay: Set marker bit inside RTP packet too
At the end of a talk spurt, not only set the marker flag on the
GstBuffer, but also set the bit inside the RTP header as recommended
by the RFC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1124>
2021-10-12 17:18:19 -04:00

427 lines
13 KiB
C

/*
* Opus Payloader Gst Element
*
* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpopuspay
* @title: rtpopuspay
*
* rtpopuspay encapsulates Opus-encoded audio data into RTP packets following
* the payload format described in RFC 7587.
*
* 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
* 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:
*
* |[
* a=rtpmap:96 multiopus/48000/6
* a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
* ]|
*
* See https://webrtc-review.googlesource.com/c/src/+/129768 for more details on
* multichannel Opus in libwebrtc.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtpopuspay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
#define GST_CAT_DEFAULT (rtpopuspay_debug)
enum
{
PROP_0,
PROP_DTX,
};
#define DEFAULT_DTX FALSE
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, 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 =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"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\" }")
);
static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopuspay, "rtpopuspay",
GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY, rtp_element_init (plugin));
#define GST_RTP_OPUS_PAY_CAST(obj) ((GstRtpOPUSPay *)(obj))
static void
gst_rtp_opus_pay_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
switch (prop_id) {
case PROP_DTX:
self->dtx = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_opus_pay_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (object);
switch (prop_id) {
case PROP_DTX:
g_value_set_boolean (value, self->dtx);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_opus_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpOPUSPay *self = GST_RTP_OPUS_PAY (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
self->marker = TRUE;
break;
default:
break;
}
ret =
GST_ELEMENT_CLASS (gst_rtp_opus_pay_parent_class)->change_state (element,
transition);
switch (transition) {
default:
break;
}
return ret;
}
static void
gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
{
GstRTPBasePayloadClass *gstbasertppayload_class;
GstElementClass *element_class;
GObjectClass *gobject_class;
gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
element_class = GST_ELEMENT_CLASS (klass);
gobject_class = (GObjectClass *) klass;
element_class->change_state = gst_rtp_opus_pay_change_state;
gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
gobject_class->set_property = gst_rtp_opus_pay_set_property;
gobject_class->get_property = gst_rtp_opus_pay_get_property;
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_pay_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_opus_pay_sink_template);
/**
* GstRtpOPUSPay:dtx:
*
* If enabled, the payloader will not transmit empty packets.
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_DTX,
g_param_spec_boolean ("dtx", "Discontinuous Transmission",
"If enabled, the payloader will not transmit empty packets",
DEFAULT_DTX,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_PLAYING |
G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (element_class,
"RTP Opus payloader",
"Codec/Payloader/Network/RTP",
"Puts Opus audio in RTP packets",
"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
"Opus RTP Payloader");
}
static void
gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
{
rtpopuspay->dtx = DEFAULT_DTX;
}
static gboolean
gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
GstCaps *src_caps;
GstStructure *s, *outcaps;
const char *encoding_name = "OPUS";
gint channels = 2;
gint rate;
gchar *encoding_params;
outcaps = gst_structure_new_empty ("unused");
src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
if (src_caps) {
GstStructure *s;
const GValue *value;
s = gst_caps_get_structure (src_caps, 0);
if (gst_structure_has_field (s, "encoding-name")) {
GValue default_value = G_VALUE_INIT;
g_value_init (&default_value, G_TYPE_STRING);
g_value_set_static_string (&default_value, encoding_name);
value = gst_structure_get_value (s, "encoding-name");
if (!gst_value_can_intersect (&default_value, value))
encoding_name = "X-GST-OPUS-DRAFT-SPITTKA-00";
}
gst_caps_unref (src_caps);
}
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels)) {
if (channels > 2) {
/* 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);
gst_structure_set (outcaps, "sprop-maxcapturerate", G_TYPE_STRING,
sprop_maxcapturerate, NULL);
g_free (sprop_maxcapturerate);
}
gst_rtp_base_payload_set_options (payload, "audio", FALSE,
encoding_name, 48000);
res = gst_rtp_base_payload_set_outcaps_structure (payload, outcaps);
gst_structure_free (outcaps);
return res;
}
static GstFlowReturn
gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpOPUSPay *self = GST_RTP_OPUS_PAY_CAST (basepayload);
GstBuffer *outbuf;
GstClockTime pts, dts, duration;
/* DTX packets are zero-length frames, with a 1 or 2-bytes header */
if (self->dtx && gst_buffer_get_size (buffer) <= 2) {
GST_LOG_OBJECT (self,
"discard empty buffer as DTX is enabled: %" GST_PTR_FORMAT, buffer);
self->marker = TRUE;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
pts = GST_BUFFER_PTS (buffer);
dts = GST_BUFFER_DTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
outbuf = gst_buffer_append (outbuf, buffer);
GST_BUFFER_PTS (outbuf) = pts;
GST_BUFFER_DTS (outbuf) = dts;
GST_BUFFER_DURATION (outbuf) = duration;
if (self->marker) {
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
self->marker = FALSE;
}
/* Push out */
return gst_rtp_base_payload_push (basepayload, outbuf);
}
static GstCaps *
gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter)
{
GstCaps *caps, *peercaps, *tcaps;
GstStructure *s;
const gchar *stereo;
if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
tcaps);
gst_caps_unref (tcaps);
if (!peercaps)
return
GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
(payload, pad, filter);
if (gst_caps_is_empty (peercaps))
return peercaps;
caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
s = gst_caps_get_structure (peercaps, 0);
stereo = gst_structure_get_string (s, "stereo");
if (stereo != NULL) {
caps = gst_caps_make_writable (caps);
if (!strcmp (stereo, "1")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
caps = gst_caps_merge (caps, caps2);
} else if (!strcmp (stereo, "0")) {
GstCaps *caps2 = gst_caps_copy (caps);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
caps = gst_caps_merge (caps, caps2);
}
}
gst_caps_unref (peercaps);
if (filter) {
GstCaps *tmp = gst_caps_intersect_full (caps, filter,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
return caps;
}