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240 lines
6.9 KiB
C
240 lines
6.9 KiB
C
/* GStreamer
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* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
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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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/**
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* SECTION:element-rtpbvpay
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* @title: rtpbvpay
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* @see_also: rtpbvdepay
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*
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* Payload BroadcomVoice audio into RTP packets according to RFC 4298.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt
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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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpbvpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpbvpay_debug);
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#define GST_CAT_DEFAULT (rtpbvpay_debug)
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static GstStaticPadTemplate gst_rtp_bv_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 ("audio/x-bv, " "mode = (int) {16, 32}")
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);
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static GstStaticPadTemplate gst_rtp_bv_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) 8000, "
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"encoding-name = (string) \"BV16\";"
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"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) 16000, " "encoding-name = (string) \"BV32\"")
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);
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static GstCaps *gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * payload,
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GstPad * pad, GstCaps * filter);
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static gboolean gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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#define gst_rtp_bv_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPBVPay, gst_rtp_bv_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
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static void
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gst_rtp_bv_pay_class_init (GstRTPBVPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpbvpay_debug, "rtpbvpay", 0,
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"BroadcomVoice audio RTP payloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_bv_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_bv_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class, "RTP BV Payloader",
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"Codec/Payloader/Network/RTP",
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"Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)",
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"Wim Taymans <wim.taymans@collabora.co.uk>");
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gstrtpbasepayload_class->set_caps = gst_rtp_bv_pay_sink_setcaps;
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gstrtpbasepayload_class->get_caps = gst_rtp_bv_pay_sink_getcaps;
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}
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static void
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gst_rtp_bv_pay_init (GstRTPBVPay * rtpbvpay)
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{
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbvpay);
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rtpbvpay->mode = -1;
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/* tell rtpbaseaudiopayload that this is a frame based codec */
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gst_rtp_base_audio_payload_set_frame_based (rtpbaseaudiopayload);
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}
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static gboolean
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gst_rtp_bv_pay_sink_setcaps (GstRTPBasePayload * rtpbasepayload, GstCaps * caps)
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{
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GstRTPBVPay *rtpbvpay;
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GstRTPBaseAudioPayload *rtpbaseaudiopayload;
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gint mode;
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GstStructure *structure;
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const char *payload_name;
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rtpbvpay = GST_RTP_BV_PAY (rtpbasepayload);
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rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpbasepayload);
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structure = gst_caps_get_structure (caps, 0);
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payload_name = gst_structure_get_name (structure);
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if (g_ascii_strcasecmp ("audio/x-bv", payload_name))
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goto wrong_caps;
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if (!gst_structure_get_int (structure, "mode", &mode))
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goto no_mode;
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if (mode != 16 && mode != 32)
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goto wrong_mode;
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if (mode == 16) {
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gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV16",
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8000);
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rtpbasepayload->clock_rate = 8000;
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} else {
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gst_rtp_base_payload_set_options (rtpbasepayload, "audio", TRUE, "BV32",
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16000);
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rtpbasepayload->clock_rate = 16000;
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}
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/* set options for this frame based audio codec */
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gst_rtp_base_audio_payload_set_frame_options (rtpbaseaudiopayload,
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mode, mode == 16 ? 10 : 20);
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if (mode != rtpbvpay->mode && rtpbvpay->mode != -1)
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goto mode_changed;
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rtpbvpay->mode = mode;
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return TRUE;
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/* ERRORS */
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wrong_caps:
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{
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GST_ERROR_OBJECT (rtpbvpay, "expected audio/x-bv, received %s",
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payload_name);
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return FALSE;
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}
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no_mode:
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{
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GST_ERROR_OBJECT (rtpbvpay, "did not receive a mode");
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return FALSE;
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}
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wrong_mode:
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{
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GST_ERROR_OBJECT (rtpbvpay, "mode must be 16 or 32, received %d", mode);
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return FALSE;
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}
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mode_changed:
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{
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GST_ERROR_OBJECT (rtpbvpay, "Mode has changed from %d to %d! "
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"Mode cannot change while streaming", rtpbvpay->mode, mode);
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return FALSE;
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}
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}
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/* we return the padtemplate caps with the mode field fixated to a value if we
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* can */
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static GstCaps *
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gst_rtp_bv_pay_sink_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
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GstCaps * filter)
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{
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GstCaps *otherpadcaps;
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GstCaps *caps;
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caps = gst_pad_get_pad_template_caps (pad);
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otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
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if (otherpadcaps) {
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if (!gst_caps_is_empty (otherpadcaps)) {
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GstStructure *structure;
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const gchar *mode_str;
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gint mode;
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structure = gst_caps_get_structure (otherpadcaps, 0);
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/* construct mode, if we can */
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mode_str = gst_structure_get_string (structure, "encoding-name");
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if (mode_str) {
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if (!strcmp (mode_str, "BV16"))
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mode = 16;
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else if (!strcmp (mode_str, "BV32"))
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mode = 32;
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else
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mode = -1;
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if (mode == 16 || mode == 32) {
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caps = gst_caps_make_writable (caps);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_set (structure, "mode", G_TYPE_INT, mode, NULL);
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}
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}
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}
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gst_caps_unref (otherpadcaps);
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}
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if (filter) {
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GstCaps *tmp;
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GST_DEBUG_OBJECT (rtppayload, "Intersect %" GST_PTR_FORMAT " and filter %"
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GST_PTR_FORMAT, caps, filter);
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tmp = gst_caps_intersect_full (filter, caps, 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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return caps;
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}
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gboolean
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gst_rtp_bv_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpbvpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_BV_PAY);
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}
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