mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
We had our own copies of those while the code was in -bad, but now we can use the existing utility functions instead of re-implementing them.
255 lines
7.7 KiB
C
255 lines
7.7 KiB
C
/*
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* Opus Payloader 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 <gst/rtp/gstrtpbuffer.h>
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#include <gst/audio/audio.h>
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#include "gstrtpopuspay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);
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#define GST_CAT_DEFAULT (rtpopuspay_debug)
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static GstStaticPadTemplate gst_rtp_opus_pay_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
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("audio/x-opus, channels = (int) [1, 2], channel-mapping-family = (int) 0")
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);
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static GstStaticPadTemplate gst_rtp_opus_pay_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 ("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-params = (string) \"2\", "
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"encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")
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);
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static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
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GstPad * pad, GstCaps * filter);
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static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)
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{
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GstRTPBasePayloadClass *gstbasertppayload_class;
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GstElementClass *element_class;
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gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;
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element_class = GST_ELEMENT_CLASS (klass);
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gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;
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gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_pay_src_template);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_opus_pay_sink_template);
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gst_element_class_set_static_metadata (element_class,
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"RTP Opus payloader",
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"Codec/Payloader/Network/RTP",
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"Puts Opus audio in RTP packets",
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"Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");
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GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,
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"Opus RTP Payloader");
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}
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static void
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gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)
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{
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}
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static gboolean
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gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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GstCaps *src_caps;
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GstStructure *s;
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char *encoding_name;
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gint channels, rate;
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const char *sprop_stereo = NULL;
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char *sprop_maxcapturerate = NULL;
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src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
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if (src_caps) {
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src_caps = gst_caps_make_writable (src_caps);
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src_caps = gst_caps_truncate (src_caps);
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s = gst_caps_get_structure (src_caps, 0);
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gst_structure_fixate_field_string (s, "encoding-name", "OPUS");
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encoding_name = g_strdup (gst_structure_get_string (s, "encoding-name"));
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gst_caps_unref (src_caps);
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} else {
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encoding_name = g_strdup ("X-GST-OPUS-DRAFT-SPITTKA-00");
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}
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s = gst_caps_get_structure (caps, 0);
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if (gst_structure_get_int (s, "channels", &channels)) {
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if (channels > 2) {
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GST_ERROR_OBJECT (payload,
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"More than 2 channels with channel-mapping-family=0 is invalid");
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return FALSE;
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} else if (channels == 2) {
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sprop_stereo = "1";
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} else {
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sprop_stereo = "0";
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}
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}
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if (gst_structure_get_int (s, "rate", &rate)) {
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sprop_maxcapturerate = g_strdup_printf ("%d", rate);
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}
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gst_rtp_base_payload_set_options (payload, "audio", FALSE,
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encoding_name, 48000);
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g_free (encoding_name);
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if (sprop_maxcapturerate && sprop_stereo) {
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res =
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gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
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G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,
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sprop_stereo, NULL);
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} else if (sprop_maxcapturerate) {
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res =
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gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",
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G_TYPE_STRING, sprop_maxcapturerate, NULL);
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} else if (sprop_stereo) {
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res =
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gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",
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G_TYPE_STRING, sprop_stereo, NULL);
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} else {
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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}
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g_free (sprop_maxcapturerate);
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return res;
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}
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static GstFlowReturn
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gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstBuffer *outbuf;
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GstClockTime pts, dts, duration;
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pts = GST_BUFFER_PTS (buffer);
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dts = GST_BUFFER_DTS (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
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gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
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outbuf = gst_buffer_append (outbuf, buffer);
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GST_BUFFER_PTS (outbuf) = pts;
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GST_BUFFER_DTS (outbuf) = dts;
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GST_BUFFER_DURATION (outbuf) = duration;
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/* Push out */
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return gst_rtp_base_payload_push (basepayload, outbuf);
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}
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static GstCaps *
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gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,
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GstPad * pad, GstCaps * filter)
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{
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GstCaps *caps, *peercaps, *tcaps;
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GstStructure *s;
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const gchar *stereo;
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if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))
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return
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GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
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(payload, pad, filter);
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tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
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peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),
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tcaps);
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gst_caps_unref (tcaps);
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if (!peercaps)
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return
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GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps
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(payload, pad, filter);
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if (gst_caps_is_empty (peercaps))
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return peercaps;
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caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));
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s = gst_caps_get_structure (peercaps, 0);
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stereo = gst_structure_get_string (s, "stereo");
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if (stereo != NULL) {
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caps = gst_caps_make_writable (caps);
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if (!strcmp (stereo, "1")) {
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GstCaps *caps2 = gst_caps_copy (caps);
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
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gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);
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caps = gst_caps_merge (caps, caps2);
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} else if (!strcmp (stereo, "0")) {
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GstCaps *caps2 = gst_caps_copy (caps);
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
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gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);
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caps = gst_caps_merge (caps, caps2);
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}
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}
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gst_caps_unref (peercaps);
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if (filter) {
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GstCaps *tmp = gst_caps_intersect_full (caps, filter,
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GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = tmp;
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}
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GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);
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return caps;
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}
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gboolean
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gst_rtp_opus_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpopuspay",
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GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY);
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}
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