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

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/* GStreamer
* Copyright (C) <2009> 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.
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*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpceltdepay.h"
#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpceltdepay_debug);
#define GST_CAT_DEFAULT (rtpceltdepay_debug)
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/* RtpCELTDepay signals and args */
#define DEFAULT_FRAMESIZE 480
#define DEFAULT_CHANNELS 1
#define DEFAULT_CLOCKRATE 32000
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enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
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};
static GstStaticPadTemplate gst_rtp_celt_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) [32000, 48000], "
"encoding-name = (string) \"CELT\"")
);
static GstStaticPadTemplate gst_rtp_celt_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-celt")
);
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static GstBuffer *gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
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static gboolean gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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#define gst_rtp_celt_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpCELTDepay, gst_rtp_celt_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpceltdepay, "rtpceltdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY, rtp_element_init (plugin));
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static void
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gst_rtp_celt_depay_class_init (GstRtpCELTDepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpceltdepay_debug, "rtpceltdepay", 0,
"CELT RTP Depayloader");
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_celt_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_celt_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP CELT depayloader", "Codec/Depayloader/Network/RTP",
"Extracts CELT audio from RTP packets",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_celt_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_celt_depay_setcaps;
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}
static void
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gst_rtp_celt_depay_init (GstRtpCELTDepay * rtpceltdepay)
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{
}
/* len 4 bytes LE,
* vendor string (len bytes),
* user_len 4 (0) bytes LE
*/
static const gchar gst_rtp_celt_comment[] =
"\045\0\0\0Depayloaded with GStreamer celtdepay\0\0\0\0";
static gboolean
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gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
GstStructure *structure;
GstRtpCELTDepay *rtpceltdepay;
gint clock_rate, nb_channels = 0, frame_size = 0;
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GstBuffer *buf;
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GstMapInfo map;
guint8 *ptr;
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const gchar *params;
GstCaps *srccaps;
gboolean res;
rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
goto no_clockrate;
depayload->clock_rate = clock_rate;
if ((params = gst_structure_get_string (structure, "encoding-params")))
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nb_channels = atoi (params);
if (!nb_channels)
nb_channels = DEFAULT_CHANNELS;
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if ((params = gst_structure_get_string (structure, "frame-size")))
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frame_size = atoi (params);
if (!frame_size)
frame_size = DEFAULT_FRAMESIZE;
rtpceltdepay->frame_size = frame_size;
GST_DEBUG_OBJECT (depayload, "clock-rate=%d channels=%d frame-size=%d",
clock_rate, nb_channels, frame_size);
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/* construct minimal header and comment packet for the decoder */
buf = gst_buffer_new_and_alloc (60);
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
ptr = map.data;
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memcpy (ptr, "CELT ", 8);
ptr += 8;
memcpy (ptr, "1.1.12", 7);
ptr += 20;
GST_WRITE_UINT32_LE (ptr, 0x80000006); /* version */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, 56); /* header_size */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, clock_rate); /* rate */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, nb_channels); /* channels */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, frame_size); /* frame-size */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, -1); /* overlap */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, -1); /* bytes_per_packet */
ptr += 4;
GST_WRITE_UINT32_LE (ptr, 0); /* extra headers */
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gst_buffer_unmap (buf, &map);
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srccaps = gst_caps_new_empty_simple ("audio/x-celt");
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
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buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
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gst_buffer_fill (buf, 0, gst_rtp_celt_comment, sizeof (gst_rtp_celt_comment));
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
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return res;
/* ERRORS */
no_clockrate:
{
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
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return FALSE;
}
}
static GstBuffer *
gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
GstBuffer *outbuf = NULL;
guint8 *payload;
guint offset, pos, payload_len, total_size, size;
guint8 s;
gint clock_rate = 0, frame_size = 0;
GstClockTime framesize_ns = 0, timestamp;
guint n = 0;
GstRtpCELTDepay *rtpceltdepay;
rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
clock_rate = depayload->clock_rate;
frame_size = rtpceltdepay->frame_size;
framesize_ns = gst_util_uint64_scale_int (frame_size, GST_SECOND, clock_rate);
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timestamp = GST_BUFFER_PTS (rtp->buffer);
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GST_LOG_OBJECT (depayload,
"got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
gst_buffer_get_size (rtp->buffer), gst_rtp_buffer_get_marker (rtp),
gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
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GST_LOG_OBJECT (depayload, "got clock-rate=%d, frame_size=%d, "
"_ns=%" GST_TIME_FORMAT ", timestamp=%" GST_TIME_FORMAT, clock_rate,
frame_size, GST_TIME_ARGS (framesize_ns), GST_TIME_ARGS (timestamp));
payload = gst_rtp_buffer_get_payload (rtp);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
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/* first count how many bytes are consumed by the size headers and make offset
* point to the first data byte */
total_size = 0;
offset = 0;
while (total_size < payload_len) {
do {
s = payload[offset++];
total_size += s + 1;
} while (s == 0xff);
}
/* offset is now pointing to the payload */
total_size = 0;
pos = 0;
while (total_size < payload_len) {
n++;
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size = 0;
do {
s = payload[pos++];
size += s;
total_size += s + 1;
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} while (s == 0xff);
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, size);
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offset += size;
if (frame_size != -1 && clock_rate != -1) {
GST_BUFFER_PTS (outbuf) = timestamp + framesize_ns * n;
GST_BUFFER_DURATION (outbuf) = framesize_ns;
}
GST_LOG_OBJECT (depayload, "push timestamp=%"
GST_TIME_FORMAT ", duration=%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
gst_rtp_drop_non_audio_meta (depayload, outbuf);
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gst_rtp_base_depayload_push (depayload, outbuf);
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}
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return NULL;
}