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0ed51294e0
This matches 7587, section 6.1: > sprop-maxcapturerate: a hint about the maximum input sampling rate > [...] > bandwidths (Table 1). By default, the sender is assumed to have > no limitations, i.e., 48000. Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4151>
263 lines
8.6 KiB
C
263 lines
8.6 KiB
C
/*
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* Opus Depayloader Gst Element
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*
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* @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpopusdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug);
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#define GST_CAT_DEFAULT (rtpopusdepay_debug)
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static GstStaticPadTemplate gst_rtp_opus_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ","
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"clock-rate = (int) 48000, "
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"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\", \"MULTIOPUS\" }")
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);
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static GstStaticPadTemplate gst_rtp_opus_depay_src_template =
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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, channel-mapping-family = (int) [ 0, 1 ]")
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);
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static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp_buffer);
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static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpopusdepay, "rtpopusdepay",
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GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass)
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{
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GstRTPBaseDepayloadClass *gstbasertpdepayload_class;
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GstElementClass *element_class;
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element_class = GST_ELEMENT_CLASS (klass);
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gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_depay_src_template);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_depay_sink_template);
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gst_element_class_set_static_metadata (element_class,
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"RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts Opus audio from RTP packets",
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"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
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gstbasertpdepayload_class->process_rtp_packet = gst_rtp_opus_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0,
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"Opus RTP Depayloader");
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}
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static void
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gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay)
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{
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}
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static gboolean
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gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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GstStructure *s;
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gboolean ret;
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const gchar *sprop_maxcapturerate;
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/* Default unless overridden by sprop_maxcapturerate */
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gint rate = 48000;
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srccaps = gst_caps_new_empty_simple ("audio/x-opus");
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s = gst_caps_get_structure (caps, 0);
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if (g_str_equal (gst_structure_get_string (s, "encoding-name"), "MULTIOPUS")) {
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gint channels;
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gint stream_count;
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gint coupled_count;
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const gchar *encoding_params;
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const gchar *num_streams;
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const gchar *coupled_streams;
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const gchar *channel_mapping;
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gchar *endptr;
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if (!gst_structure_has_field_typed (s, "encoding-params", G_TYPE_STRING) ||
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!gst_structure_has_field_typed (s, "num_streams", G_TYPE_STRING) ||
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!gst_structure_has_field_typed (s, "coupled_streams", G_TYPE_STRING) ||
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!gst_structure_has_field_typed (s, "channel_mapping", G_TYPE_STRING)) {
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GST_WARNING_OBJECT (depayload, "Encoding name 'MULTIOPUS' requires "
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"encoding-params, num_streams, coupled_streams and channel_mapping "
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"as string fields in caps.");
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goto reject_caps;
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}
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gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 1,
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NULL);
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encoding_params = gst_structure_get_string (s, "encoding-params");
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channels = g_ascii_strtoull (encoding_params, &endptr, 10);
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if (*endptr != '\0' || channels > 255) {
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GST_WARNING_OBJECT (depayload, "Invalid encoding-params value '%s'",
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encoding_params);
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goto reject_caps;
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}
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gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, channels, NULL);
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num_streams = gst_structure_get_string (s, "num_streams");
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stream_count = g_ascii_strtoull (num_streams, &endptr, 10);
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if (*endptr != '\0' || stream_count > channels) {
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GST_WARNING_OBJECT (depayload, "Invalid num_streams value '%s'",
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num_streams);
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goto reject_caps;
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}
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gst_caps_set_simple (srccaps, "stream-count", G_TYPE_INT, stream_count,
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NULL);
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coupled_streams = gst_structure_get_string (s, "coupled_streams");
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coupled_count = g_ascii_strtoull (coupled_streams, &endptr, 10);
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if (*endptr != '\0' || coupled_count > stream_count) {
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GST_WARNING_OBJECT (depayload, "Invalid coupled_streams value '%s'",
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coupled_streams);
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goto reject_caps;
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}
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gst_caps_set_simple (srccaps, "coupled-count", G_TYPE_INT, coupled_count,
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NULL);
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channel_mapping = gst_structure_get_string (s, "channel_mapping");
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{
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gchar **split;
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gchar **ptr;
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GValue mapping = G_VALUE_INIT;
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GValue v = G_VALUE_INIT;
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split = g_strsplit (channel_mapping, ",", -1);
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g_value_init (&mapping, GST_TYPE_ARRAY);
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g_value_init (&v, G_TYPE_INT);
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for (ptr = split; *ptr; ++ptr) {
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gint channel = g_ascii_strtoull (*ptr, &endptr, 10);
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if (*endptr != '\0' || channel > channels) {
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GST_WARNING_OBJECT (depayload, "Invalid channel_mapping value '%s'",
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channel_mapping);
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g_value_unset (&mapping);
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break;
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}
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g_value_set_int (&v, channel);
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gst_value_array_append_value (&mapping, &v);
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}
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g_value_unset (&v);
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g_strfreev (split);
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if (G_IS_VALUE (&mapping)) {
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gst_caps_set_value (srccaps, "channel-mapping", &mapping);
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g_value_unset (&mapping);
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} else {
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goto reject_caps;
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}
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}
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} else {
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const gchar *sprop_stereo;
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gst_caps_set_simple (srccaps, "channel-mapping-family", G_TYPE_INT, 0,
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NULL);
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if ((sprop_stereo = gst_structure_get_string (s, "sprop-stereo"))) {
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if (strcmp (sprop_stereo, "0") == 0)
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gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 1, NULL);
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else if (strcmp (sprop_stereo, "1") == 0)
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gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL);
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else
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GST_WARNING_OBJECT (depayload, "Unknown sprop-stereo value '%s'",
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sprop_stereo);
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} else {
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/* Although sprop-stereo defaults to mono as per RFC 7587, this just means
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that the signal is likely mono and can be safely downmixed, it may
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still be stereo at times. */
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gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL);
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}
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}
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if ((sprop_maxcapturerate =
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gst_structure_get_string (s, "sprop-maxcapturerate"))) {
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gchar *tailptr;
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gulong tmp_rate;
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tmp_rate = strtoul (sprop_maxcapturerate, &tailptr, 10);
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if (tmp_rate > INT_MAX || *tailptr != '\0') {
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GST_WARNING_OBJECT (depayload,
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"Failed to parse sprop-maxcapturerate value '%s'",
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sprop_maxcapturerate);
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} else {
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/* Valid rate from sprop, let's use it */
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rate = tmp_rate;
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}
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}
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gst_caps_set_simple (srccaps, "rate", G_TYPE_INT, rate, NULL);
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ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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GST_DEBUG_OBJECT (depayload,
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"set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret);
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gst_caps_unref (srccaps);
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depayload->clock_rate = 48000;
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return ret;
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reject_caps:
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gst_caps_unref (srccaps);
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return FALSE;
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}
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static GstBuffer *
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gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp_buffer)
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{
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GstBuffer *outbuf;
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp_buffer);
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gst_rtp_drop_non_audio_meta (depayload, outbuf);
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return outbuf;
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}
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