gstreamer/gst/rtp/gstrtpqcelpdepay.c
Tim-Philipp Müller 4a28e649c3 rtp: cache meta tag quarks and add more utility functions for metas
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.
2017-05-24 13:32:10 +01:00

432 lines
11 KiB
C

/* GStreamer
* Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpqcelpdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpqcelpdepay_debug);
#define GST_CAT_DEFAULT (rtpqcelpdepay_debug)
/* references:
*
* RFC 2658 - RTP Payload Format for PureVoice(tm) Audio
*/
#define FRAME_DURATION (20 * GST_MSECOND)
/* RtpQCELPDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
static GstStaticPadTemplate gst_rtp_qcelp_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"QCELP\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_QCELP_STRING ", "
"clock-rate = (int) 8000")
);
static GstStaticPadTemplate gst_rtp_qcelp_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/qcelp, " "channels = (int) 1," "rate = (int) 8000")
);
static void gst_rtp_qcelp_depay_finalize (GObject * object);
static gboolean gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
#define gst_rtp_qcelp_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpQCELPDepay, gst_rtp_qcelp_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void
gst_rtp_qcelp_depay_class_init (GstRtpQCELPDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gobject_class->finalize = gst_rtp_qcelp_depay_finalize;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_qcelp_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_qcelp_depay_setcaps;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_qcelp_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_qcelp_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP QCELP depayloader", "Codec/Depayloader/Network/RTP",
"Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpqcelpdepay_debug, "rtpqcelpdepay", 0,
"QCELP RTP Depayloader");
}
static void
gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay)
{
}
static void
gst_rtp_qcelp_depay_finalize (GObject * object)
{
GstRtpQCELPDepay *depay;
depay = GST_RTP_QCELP_DEPAY (object);
if (depay->packets != NULL) {
g_ptr_array_foreach (depay->packets, (GFunc) gst_buffer_unref, NULL);
g_ptr_array_free (depay->packets, TRUE);
depay->packets = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_qcelp_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean res;
srccaps = gst_caps_new_simple ("audio/qcelp",
"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);
res = gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return res;
}
static const gint frame_size[16] = {
1, 4, 8, 17, 35, -8, 0, 0,
0, 0, 0, 0, 0, 0, 1, 0
};
/* get the frame length, 0 is invalid, negative values are invalid but can be
* recovered from. */
static gint
get_frame_len (GstRtpQCELPDepay * depay, guint8 frame_type)
{
if (frame_type >= G_N_ELEMENTS (frame_size))
return 0;
return frame_size[frame_type];
}
static guint
count_packets (GstRtpQCELPDepay * depay, guint8 * data, guint size)
{
guint count = 0;
while (size > 0) {
gint frame_len;
frame_len = get_frame_len (depay, data[0]);
/* 0 is invalid and we throw away the remainder of the frames */
if (frame_len == 0)
break;
if (frame_len < 0)
frame_len = -frame_len;
if (frame_len > size)
break;
size -= frame_len;
data += frame_len;
count++;
}
return count;
}
static void
flush_packets (GstRtpQCELPDepay * depay)
{
guint i, size;
GST_DEBUG_OBJECT (depay, "flushing packets");
size = depay->packets->len;
for (i = 0; i < size; i++) {
GstBuffer *outbuf;
outbuf = g_ptr_array_index (depay->packets, i);
g_ptr_array_index (depay->packets, i) = NULL;
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (depay), outbuf);
}
/* and reset interleaving state */
depay->interleaved = FALSE;
depay->bundling = 0;
}
static void
add_packet (GstRtpQCELPDepay * depay, guint LLL, guint NNN, guint index,
GstBuffer * outbuf)
{
guint idx;
GstBuffer *old;
/* figure out the position in the array, note that index is never 0 because we
* push those packets immediately. */
idx = NNN + ((LLL + 1) * (index - 1));
GST_DEBUG_OBJECT (depay, "adding packet at index %u", idx);
/* free old buffer (should not happen) */
old = g_ptr_array_index (depay->packets, idx);
if (old)
gst_buffer_unref (old);
/* store new buffer */
g_ptr_array_index (depay->packets, idx) = outbuf;
}
static GstBuffer *
create_erasure_buffer (GstRtpQCELPDepay * depay)
{
GstBuffer *outbuf;
GstMapInfo map;
outbuf = gst_buffer_new_and_alloc (1);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
map.data[0] = 14;
gst_buffer_unmap (outbuf, &map);
return outbuf;
}
static GstBuffer *
gst_rtp_qcelp_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp)
{
GstRtpQCELPDepay *depay;
GstBuffer *outbuf;
GstClockTime timestamp;
guint payload_len, offset, index;
guint8 *payload;
guint LLL, NNN;
depay = GST_RTP_QCELP_DEPAY (depayload);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
if (payload_len < 2)
goto too_small;
timestamp = GST_BUFFER_PTS (rtp->buffer);
payload = gst_rtp_buffer_get_payload (rtp);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |RR | LLL | NNN |
* +-+-+-+-+-+-+-+-+
*/
/* RR = payload[0] >> 6; */
LLL = (payload[0] & 0x38) >> 3;
NNN = (payload[0] & 0x07);
payload_len--;
payload++;
GST_DEBUG_OBJECT (depay, "LLL %u, NNN %u", LLL, NNN);
if (LLL > 5)
goto invalid_lll;
if (NNN > LLL)
goto invalid_nnn;
if (LLL != 0) {
/* we are interleaved */
if (!depay->interleaved) {
guint size;
GST_DEBUG_OBJECT (depay, "starting interleaving group");
/* bundling is not allowed to change in one interleave group */
depay->bundling = count_packets (depay, payload, payload_len);
GST_DEBUG_OBJECT (depay, "got bundling of %u", depay->bundling);
/* we have one bundle where NNN goes from 0 to L, we don't store the index
* 0 frames, so L+1 packets. Each packet has 'bundling - 1' packets */
size = (depay->bundling - 1) * (LLL + 1);
/* create the array to hold the packets */
if (depay->packets == NULL)
depay->packets = g_ptr_array_sized_new (size);
GST_DEBUG_OBJECT (depay, "created packet array of size %u", size);
g_ptr_array_set_size (depay->packets, size);
/* we were previously not interleaved, figure out how much space we
* need to deinterleave */
depay->interleaved = TRUE;
}
} else {
/* we are not interleaved */
if (depay->interleaved) {
GST_DEBUG_OBJECT (depay, "stopping interleaving");
/* flush packets if we were previously interleaved */
flush_packets (depay);
}
depay->bundling = 0;
}
index = 0;
offset = 1;
while (payload_len > 0) {
gint frame_len;
gboolean do_erasure;
frame_len = get_frame_len (depay, payload[0]);
GST_DEBUG_OBJECT (depay, "got frame len %d", frame_len);
if (frame_len == 0)
goto invalid_frame;
if (frame_len < 0) {
/* need to add an erasure frame but we can recover */
frame_len = -frame_len;
do_erasure = TRUE;
} else {
do_erasure = FALSE;
}
if (frame_len > payload_len)
goto invalid_frame;
if (do_erasure) {
/* create erasure frame */
outbuf = create_erasure_buffer (depay);
} else {
/* each frame goes into its buffer */
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, frame_len);
}
GST_BUFFER_PTS (outbuf) = timestamp;
GST_BUFFER_DURATION (outbuf) = FRAME_DURATION;
gst_rtp_drop_non_audio_meta (depayload, outbuf);
if (!depay->interleaved || index == 0) {
/* not interleaved or first frame in packet, just push */
gst_rtp_base_depayload_push (depayload, outbuf);
if (timestamp != -1)
timestamp += FRAME_DURATION;
} else {
/* put in interleave buffer */
add_packet (depay, LLL, NNN, index, outbuf);
if (timestamp != -1)
timestamp += (FRAME_DURATION * (LLL + 1));
}
payload_len -= frame_len;
payload += frame_len;
offset += frame_len;
index++;
/* discard excess packets */
if (depay->bundling > 0 && depay->bundling <= index)
break;
}
while (index < depay->bundling) {
GST_DEBUG_OBJECT (depay, "filling with erasure buffer");
/* fill remainder with erasure packets */
outbuf = create_erasure_buffer (depay);
add_packet (depay, LLL, NNN, index, outbuf);
index++;
}
if (depay->interleaved && LLL == NNN) {
GST_DEBUG_OBJECT (depay, "interleave group ended, flushing");
/* we have the complete interleave group, flush */
flush_packets (depay);
}
return NULL;
/* ERRORS */
too_small:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP payload too small (%d)", payload_len));
return NULL;
}
invalid_lll:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid LLL received (%d)", LLL));
return NULL;
}
invalid_nnn:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid NNN received (%d)", NNN));
return NULL;
}
invalid_frame:
{
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
(NULL), ("QCELP RTP invalid frame received"));
return NULL;
}
}
gboolean
gst_rtp_qcelp_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpqcelpdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY);
}