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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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347 lines
9.3 KiB
C
347 lines
9.3 KiB
C
/* GStreamer
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* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
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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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpspeexpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
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#define GST_CAT_DEFAULT (rtpspeexpay_debug)
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static GstStaticPadTemplate gst_rtp_speex_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-speex, "
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"rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_speex_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) [ 6000, 48000 ], "
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"encoding-name = (string) \"SPEEX\", "
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"encoding-params = (string) \"1\"")
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);
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static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstCaps *gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload,
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GstPad * pad, GstCaps * filter);
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static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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#define gst_rtp_speex_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gstelement_class->change_state = gst_rtp_speex_pay_change_state;
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gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps;
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gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_speex_pay_sink_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_speex_pay_src_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Speex payloader", "Codec/Payloader/Network/RTP",
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"Payload-encodes Speex audio into a RTP packet",
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"Edgard Lima <edgard.lima@indt.org.br>");
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GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
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"Speex RTP Payloader");
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}
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static void
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gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay)
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{
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GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
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GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
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}
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static gboolean
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gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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/* don't configure yet, we wait for the ident packet */
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return TRUE;
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}
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static GstCaps *
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gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, 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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otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
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caps = gst_pad_get_pad_template_caps (pad);
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if (otherpadcaps) {
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if (!gst_caps_is_empty (otherpadcaps)) {
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GstStructure *ps;
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GstStructure *s;
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gint clock_rate;
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ps = gst_caps_get_structure (otherpadcaps, 0);
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caps = gst_caps_make_writable (caps);
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s = gst_caps_get_structure (caps, 0);
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if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
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gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
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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 *tcaps = caps;
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caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (tcaps);
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}
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return caps;
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}
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static gboolean
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gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
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const guint8 * data, guint size)
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{
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guint32 version, header_size, rate, mode, nb_channels;
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GstRTPBasePayload *payload;
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gchar *cstr;
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gboolean res;
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/* we need the header string (8), the version string (20), the version
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* and the header length. */
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if (size < 36)
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goto too_small;
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if (!g_str_has_prefix ((const gchar *) data, "Speex "))
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goto wrong_header;
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/* skip header and version string */
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data += 28;
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version = GST_READ_UINT32_LE (data);
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if (version != 1)
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goto wrong_version;
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data += 4;
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/* ensure sizes */
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header_size = GST_READ_UINT32_LE (data);
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if (header_size < 80)
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goto header_too_small;
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if (size < header_size)
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goto payload_too_small;
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data += 4;
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rate = GST_READ_UINT32_LE (data);
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data += 4;
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mode = GST_READ_UINT32_LE (data);
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data += 8;
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nb_channels = GST_READ_UINT32_LE (data);
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GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
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rate, mode, nb_channels);
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payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay);
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gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
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cstr = g_strdup_printf ("%d", nb_channels);
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res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
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G_TYPE_STRING, cstr, NULL);
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g_free (cstr);
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return res;
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/* ERRORS */
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too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"ident packet too small, need at least 32 bytes");
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return FALSE;
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}
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wrong_header:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"ident packet does not start with \"Speex \"");
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return FALSE;
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}
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wrong_version:
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{
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GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
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version);
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return FALSE;
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}
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header_too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"header size too small, need at least 80 bytes, " "got only %d",
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header_size);
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return FALSE;
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}
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payload_too_small:
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{
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GST_DEBUG_OBJECT (rtpspeexpay,
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"payload too small, need at least %d bytes, got only %d", header_size,
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size);
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpSPEEXPay *rtpspeexpay;
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GstMapInfo map;
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GstBuffer *outbuf;
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GstClockTime timestamp, duration;
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GstFlowReturn ret;
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rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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switch (rtpspeexpay->packet) {
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case 0:
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/* ident packet. We need to parse the headers to construct the RTP
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* properties. */
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if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) {
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gst_buffer_unmap (buffer, &map);
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goto parse_error;
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}
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ret = GST_FLOW_OK;
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gst_buffer_unmap (buffer, &map);
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goto done;
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case 1:
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/* comment packet, we ignore it */
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ret = GST_FLOW_OK;
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gst_buffer_unmap (buffer, &map);
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goto done;
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default:
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/* other packets go in the payload */
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break;
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}
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gst_buffer_unmap (buffer, &map);
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
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ret = GST_FLOW_OK;
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goto done;
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}
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timestamp = GST_BUFFER_PTS (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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/* FIXME, only one SPEEX frame per RTP packet for now */
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outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
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/* FIXME, assert for now */
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g_assert (gst_buffer_get_size (buffer) <=
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GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));
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/* copy timestamp and duration */
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GST_BUFFER_PTS (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = duration;
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outbuf = gst_buffer_append (outbuf, buffer);
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buffer = NULL;
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ret = gst_rtp_base_payload_push (basepayload, outbuf);
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done:
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if (buffer)
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gst_buffer_unref (buffer);
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rtpspeexpay->packet++;
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return ret;
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/* ERRORS */
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parse_error:
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{
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GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
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("Error parsing first identification packet."));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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}
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static GstStateChangeReturn
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gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpSPEEXPay *rtpspeexpay;
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GstStateChangeReturn ret;
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rtpspeexpay = GST_RTP_SPEEX_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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rtpspeexpay->packet = 0;
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpspeexpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);
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}
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