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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4a28e649c3
Every g_quark_from_static_string() is a hash table lookup serialised on the global quark lock in GLib. Let's just look up the two quarks we need once and cache them locally for future use. While we're at it, add new utility functions for the two most commonly used tags (audio + video). Make first argument a gpointer so we don't have to cast and make the code ugly. These are used for logging purposes only anyway.
350 lines
9.4 KiB
C
350 lines
9.4 KiB
C
/* GStreamer
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* Copyright (C) <2005> Edgard Lima <edgard.lima@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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#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 <gst/audio/audio.h>
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#include "gstrtpspeexpay.h"
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#include "gstrtputils.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_static_pad_template (gstelement_class,
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&gst_rtp_speex_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&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@gmail.com>");
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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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gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
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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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