diff options
Diffstat (limited to 'gst/rtp/gstrtpopusdepay.c')
| -rw-r--r-- | gst/rtp/gstrtpopusdepay.c | 175 |
1 files changed, 175 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpopusdepay.c b/gst/rtp/gstrtpopusdepay.c new file mode 100644 index 000000000..8152cd57f --- /dev/null +++ b/gst/rtp/gstrtpopusdepay.c @@ -0,0 +1,175 @@ +/* + * Opus Depayloader 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. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include <string.h> +#include <stdlib.h> +#include <gst/rtp/gstrtpbuffer.h> +#include <gst/audio/audio.h> +#include "gstrtpopusdepay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpopusdepay_debug); +#define GST_CAT_DEFAULT (rtpopusdepay_debug) + +static GstStaticPadTemplate gst_rtp_opus_depay_sink_template = +GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + 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\" }") + ); + +static GstStaticPadTemplate gst_rtp_opus_depay_src_template = +GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS ("audio/x-opus, channel-mapping-family = (int) 0") + ); + +static GstBuffer *gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, + GstBuffer * buf); +static gboolean gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, + GstCaps * caps); + +G_DEFINE_TYPE (GstRTPOpusDepay, gst_rtp_opus_depay, + GST_TYPE_RTP_BASE_DEPAYLOAD); + +static void +gst_rtp_opus_depay_class_init (GstRTPOpusDepayClass * klass) +{ + GstRTPBaseDepayloadClass *gstbasertpdepayload_class; + GstElementClass *element_class; + + element_class = GST_ELEMENT_CLASS (klass); + gstbasertpdepayload_class = (GstRTPBaseDepayloadClass *) klass; + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_opus_depay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_opus_depay_sink_template)); + gst_element_class_set_static_metadata (element_class, + "RTP Opus packet depayloader", "Codec/Depayloader/Network/RTP", + "Extracts Opus audio from RTP packets", + "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>"); + + gstbasertpdepayload_class->process = gst_rtp_opus_depay_process; + gstbasertpdepayload_class->set_caps = gst_rtp_opus_depay_setcaps; + + GST_DEBUG_CATEGORY_INIT (rtpopusdepay_debug, "rtpopusdepay", 0, + "Opus RTP Depayloader"); +} + +static void +gst_rtp_opus_depay_init (GstRTPOpusDepay * rtpopusdepay) +{ + +} + +static gboolean +gst_rtp_opus_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) +{ + GstCaps *srccaps; + GstStructure *s; + gboolean ret; + const gchar *sprop_stereo, *sprop_maxcapturerate; + + srccaps = + gst_caps_new_simple ("audio/x-opus", "channel-mapping-family", G_TYPE_INT, + 0, NULL); + + s = gst_caps_get_structure (caps, 0); + if ((sprop_stereo = gst_structure_get_string (s, "sprop-stereo"))) { + if (strcmp (sprop_stereo, "0") == 0) + gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 1, NULL); + else if (strcmp (sprop_stereo, "1") == 0) + gst_caps_set_simple (srccaps, "channels", G_TYPE_INT, 2, NULL); + else + GST_WARNING_OBJECT (depayload, "Unknown sprop-stereo value '%s'", + sprop_stereo); + } + + if ((sprop_maxcapturerate = + gst_structure_get_string (s, "sprop-maxcapturerate"))) { + gulong rate; + gchar *tailptr; + + rate = strtoul (sprop_maxcapturerate, &tailptr, 10); + if (rate > INT_MAX || *tailptr != '\0') { + GST_WARNING_OBJECT (depayload, + "Failed to parse sprop-maxcapturerate value '%s'", + sprop_maxcapturerate); + } else { + gst_caps_set_simple (srccaps, "rate", G_TYPE_INT, rate, NULL); + } + } + + ret = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps); + + GST_DEBUG_OBJECT (depayload, + "set caps on source: %" GST_PTR_FORMAT " (ret=%d)", srccaps, ret); + gst_caps_unref (srccaps); + + depayload->clock_rate = 48000; + + return ret; +} + +static gboolean +foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data) +{ + GstRTPOpusDepay *depay = user_data; + const GstMetaInfo *info = (*meta)->info; + const gchar *const *tags = gst_meta_api_type_get_tags (info->api); + + if (!tags || (g_strv_length ((gchar **) tags) == 1 + && gst_meta_api_type_has_tag (info->api, + g_quark_from_string (GST_META_TAG_AUDIO_STR)))) { + GST_DEBUG_OBJECT (depay, "keeping metadata %s", g_type_name (info->api)); + } else { + GST_DEBUG_OBJECT (depay, "dropping metadata %s", g_type_name (info->api)); + *meta = NULL; + } + + return TRUE; +} + +static GstBuffer * +gst_rtp_opus_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf) +{ + GstBuffer *outbuf; + GstRTPBuffer rtpbuf = { NULL, }; + + gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf); + outbuf = gst_rtp_buffer_get_payload_buffer (&rtpbuf); + gst_rtp_buffer_unmap (&rtpbuf); + + outbuf = gst_buffer_make_writable (outbuf); + /* Filter away all metas that are not sensible to copy */ + gst_buffer_foreach_meta (outbuf, foreach_metadata, depayload); + + return outbuf; +} |