gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpqcelpdepay.c

428 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 "gstrtpelements.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);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpqcelpdepay, "rtpqcelpdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_QCELP_DEPAY, rtp_element_init (plugin));
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;
}
}