mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Remove the pt restrictions for all the depayloaders that have an encoding-name. We can use this to autoplug decoders. Remove the encoding-name for all the payloaders with a fixed payload type. We now either have an encoding-name or a pt in the sinkpad caps of a depayloader. See https://bugzilla.gnome.org/show_bug.cgi?id=639292
438 lines
11 KiB
C
438 lines
11 KiB
C
/* GStreamer
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* Copyright (C) <2010> 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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <gst/rtp/gstrtpbuffer.h>
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#include <stdlib.h>
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#include <string.h>
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#include "gstrtpqcelpdepay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug);
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#define GST_CAT_DEFAULT (rtpqcelpdepay_debug)
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/* references:
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*
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* RFC 2658 - RTP Payload Format for PureVoice(tm) Audio
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*/
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#define FRAME_DURATION (20 * GST_MSECOND)
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/* RtpQCELPDepay signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0
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};
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static GstStaticPadTemplate gst_rtp_qcelp_depay_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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"clock-rate = (int) 8000, "
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"encoding-name = (string) \"QCELP\"; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", "
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"clock-rate = (int) 8000")
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);
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static GstStaticPadTemplate gst_rtp_qcelp_depay_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 ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000")
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);
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static void gst_rtp_qcelp_depay_finalize (GObject * object);
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static gboolean gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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#define gst_rtp_qcelp_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpQCELPDepay, gst_rtp_qcelp_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void
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gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gobject_class->finalize = gst_rtp_qcelp_depay_finalize;
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gstrtpbasedepayload_class->process = gst_rtp_qcelp_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_qcelp_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_qcelp_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP QCELP depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)",
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0,
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"QCELP RTP Depayloader");
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}
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static void
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gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay)
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{
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GstRTPBaseDepayload G_GNUC_UNUSED *depayload;
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depayload = GST_RTP_BASE_DEPAYLOAD (rtpqcelpdepay);
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}
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static void
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gst_rtp_qcelp_depay_finalize (GObject * object)
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{
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GstRtpQCELPDepay *depay;
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depay = GST_RTP_QCELP_DEPAY (object);
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if (depay->packets != NULL) {
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g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL);
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g_ptr_array_free (depay->packets, TRUE);
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depay->packets = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstCaps *srccaps;
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gboolean res;
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srccaps = gst_caps_new_simple ("audio/qcelp",
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
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res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
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gst_caps_unref (srccaps);
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return res;
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}
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static const gint frame_size[16] = {
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1, 4, 8, 17, 35, -8, 0, 0,
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0, 0, 0, 0, 0, 0, 1, 0
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};
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/* get the frame length, 0 is invalid, negative values are invalid but can be
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* recovered from. */
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static gint
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get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type)
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{
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if (frame_type >= G_N_ELEMENTS (frame_size))
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return 0;
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return frame_size[frame_type];
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}
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static guint
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count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
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{
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guint count = 0;
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while (size > 0) {
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gint frame_len;
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frame_len = get_frame_len (depay, data[0]);
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/* 0 is invalid and we throw away the remainder of the frames */
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if (frame_len == 0)
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break;
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if (frame_len < 0)
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frame_len = -frame_len;
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if (frame_len > size)
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break;
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size -= frame_len;
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data += frame_len;
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count++;
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}
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return count;
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}
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static void
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flush_packets (GstRtpQCELPDepay * depay)
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{
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guint i, size;
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GST_DEBUG_OBJECT (depay, "flushing packets");
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size = depay->packets->len;
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for (i = 0; i < size; i++) {
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GstBuffer *outbuf;
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outbuf = g_ptr_array_index (depay->packets, i);
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g_ptr_array_index (depay->packets, i) = NULL;
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (depay), outbuf);
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}
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/* and reset interleaving state */
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depay->interleaved = FALSE;
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depay->bundling = 0;
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}
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static void
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add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index,
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GstBuffer * outbuf)
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{
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guint idx;
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GstBuffer *old;
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/* figure out the position in the array, note that index is never 0 because we
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* push those packets immediately. */
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idx = NNN + ((LLL + 1) * (index - 1));
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GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx);
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/* free old buffer (should not happen) */
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old = g_ptr_array_index (depay->packets, idx);
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if (old)
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gst_buffer_unref (old);
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/* store new buffer */
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g_ptr_array_index (depay->packets, idx) = outbuf;
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}
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static GstBuffer *
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create_erasure_buffer (GstRtpQCELPDepay * depay)
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{
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GstBuffer *outbuf;
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GstMapInfo map;
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outbuf = gst_buffer_new_and_alloc (1);
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gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
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map.data[0] = 14;
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gst_buffer_unmap (outbuf, &map);
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return outbuf;
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}
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static GstBuffer *
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gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstRtpQCELPDepay *depay;
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GstBuffer *outbuf;
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GstClockTime timestamp;
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guint payload_len, offset, index;
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guint8 *payload;
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guint LLL, NNN;
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GstRTPBuffer rtp = { NULL };
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depay = GST_RTP_QCELP_DEPAY (depayload);
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
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payload_len = gst_rtp_buffer_get_payload_len (&rtp);
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if (payload_len < 2)
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goto too_small;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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payload = gst_rtp_buffer_get_payload (&rtp);
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/* 0 1 2 3 4 5 6 7
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* +-+-+-+-+-+-+-+-+
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* |RR | LLL | NNN |
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* +-+-+-+-+-+-+-+-+
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*/
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/* RR = payload[0] >> 6; */
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LLL = (payload[0] & 0x38) >> 3;
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NNN = (payload[0] & 0x07);
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payload_len--;
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payload++;
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GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);
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if (LLL > 5)
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goto invalid_lll;
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if (NNN > LLL)
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goto invalid_nnn;
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if (LLL != 0) {
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/* we are interleaved */
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if (!depay->interleaved) {
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guint size;
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GST_DEBUG_OBJECT (depay, "starting interleaving group");
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/* bundling is not allowed to change in one interleave group */
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depay->bundling = count_packets (depay, payload, payload_len);
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GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
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/* we have one bundle where NNN goes from 0 to L, we don't store the index
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* 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
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size = (depay->bundling - 1) * (LLL + 1);
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/* create the array to hold the packets */
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if (depay->packets == NULL)
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depay->packets = g_ptr_array_sized_new (size);
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GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
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g_ptr_array_set_size (depay->packets, size);
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/* we were previously not interleaved, figure out how much space we
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* need to deinterleave */
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depay->interleaved = TRUE;
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}
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} else {
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/* we are not interleaved */
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if (depay->interleaved) {
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GST_DEBUG_OBJECT (depay, "stopping interleaving");
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/* flush packets if we were previously interleaved */
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flush_packets (depay);
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}
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depay->bundling = 0;
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}
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index = 0;
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offset = 1;
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while (payload_len > 0) {
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gint frame_len;
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gboolean do_erasure;
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frame_len = get_frame_len (depay, payload[0]);
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GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);
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if (frame_len == 0)
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goto invalid_frame;
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if (frame_len < 0) {
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/* need to add an erasure frame but we can recover */
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frame_len = -frame_len;
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do_erasure = TRUE;
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} else {
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do_erasure = FALSE;
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}
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if (frame_len > payload_len)
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goto invalid_frame;
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if (do_erasure) {
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/* create erasure frame */
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outbuf = create_erasure_buffer (depay);
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} else {
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/* each frame goes into its buffer */
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outbuf = gst_rtp_buffer_get_payload_subbuffer (&rtp, offset, frame_len);
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}
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GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
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GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;
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if (!depay->interleaved || index == 0) {
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/* not interleaved or first frame in packet, just push */
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gst_rtp_base_depayload_push (depayload, outbuf);
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if (timestamp != -1)
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timestamp += FRAME_DURATION;
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} else {
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/* put in interleave buffer */
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add_packet (depay, LLL, NNN, index, outbuf);
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if (timestamp != -1)
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timestamp += (FRAME_DURATION * (LLL + 1));
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}
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payload_len -= frame_len;
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payload += frame_len;
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offset += frame_len;
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index++;
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/* discard excess packets */
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if (depay->bundling > 0 && depay->bundling <= index)
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break;
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}
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while (index < depay->bundling) {
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GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
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/* fill remainder with erasure packets */
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outbuf = create_erasure_buffer (depay);
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add_packet (depay, LLL, NNN, index, outbuf);
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index++;
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}
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if (depay->interleaved && LLL == NNN) {
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GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
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/* we have the complete interleave group, flush */
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flush_packets (depay);
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}
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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/* ERRORS */
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too_small:
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{
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GST_ELEMENT_WARNING (depay, STREAM, DECODE,
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(NULL), ("QCELP RTP payload too small (%d)", payload_len));
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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invalid_lll:
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{
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GST_ELEMENT_WARNING (depay, STREAM, DECODE,
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(NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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invalid_nnn:
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{
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GST_ELEMENT_WARNING (depay, STREAM, DECODE,
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(NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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invalid_frame:
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{
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GST_ELEMENT_WARNING (depay, STREAM, DECODE,
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(NULL), ("QCELP RTP invalid frame received"));
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gst_rtp_buffer_unmap (&rtp);
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return NULL;
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}
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}
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gboolean
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gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpqcelpdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY);
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}
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