mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
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.
285 lines
8.3 KiB
C
285 lines
8.3 KiB
C
/* GStreamer
|
|
* Copyright (C) <2007> 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.
|
|
*/
|
|
|
|
/**
|
|
* SECTION:element-rtpL16depay
|
|
* @see_also: rtpL16pay
|
|
*
|
|
* Extract raw audio from RTP packets according to RFC 3551.
|
|
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
|
|
*
|
|
* <refsect2>
|
|
* <title>Example pipeline</title>
|
|
* |[
|
|
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
|
|
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
|
|
* the rtpL16pay example to create the RTP stream.
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
#include <stdlib.h>
|
|
|
|
#include <gst/audio/audio.h>
|
|
|
|
#include "gstrtpL16depay.h"
|
|
#include "gstrtpchannels.h"
|
|
#include "gstrtputils.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
|
|
#define GST_CAT_DEFAULT (rtpL16depay_debug)
|
|
|
|
static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw, "
|
|
"format = (string) S16BE, "
|
|
"layout = (string) interleaved, "
|
|
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_L16_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) [ 1, MAX ], "
|
|
/* "channels = (int) [1, MAX]" */
|
|
/* "emphasis = (string) ANY" */
|
|
/* "channel-order = (string) ANY" */
|
|
"encoding-name = (string) \"L16\";"
|
|
"application/x-rtp, "
|
|
"media = (string) \"audio\", "
|
|
"payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
|
|
GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]"
|
|
/* "channels = (int) [1, MAX]" */
|
|
/* "emphasis = (string) ANY" */
|
|
/* "channel-order = (string) ANY" */
|
|
)
|
|
);
|
|
|
|
#define gst_rtp_L16_depay_parent_class parent_class
|
|
G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
|
|
|
|
static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
|
|
GstCaps * caps);
|
|
static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
|
|
GstRTPBuffer * rtp);
|
|
|
|
static void
|
|
gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
|
|
{
|
|
GstElementClass *gstelement_class;
|
|
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
|
|
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
|
|
|
|
gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
|
|
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
|
|
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_L16_depay_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_L16_depay_sink_template);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTP audio depayloader", "Codec/Depayloader/Network/RTP",
|
|
"Extracts raw audio from RTP packets",
|
|
"Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
|
|
"Raw Audio RTP Depayloader");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
|
|
{
|
|
}
|
|
|
|
static gint
|
|
gst_rtp_L16_depay_parse_int (GstStructure * structure, const gchar * field,
|
|
gint def)
|
|
{
|
|
const gchar *str;
|
|
gint res;
|
|
|
|
if ((str = gst_structure_get_string (structure, field)))
|
|
return atoi (str);
|
|
|
|
if (gst_structure_get_int (structure, field, &res))
|
|
return res;
|
|
|
|
return def;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
|
|
{
|
|
GstStructure *structure;
|
|
GstRtpL16Depay *rtpL16depay;
|
|
gint clock_rate, payload;
|
|
gint channels;
|
|
GstCaps *srccaps;
|
|
gboolean res;
|
|
const gchar *channel_order;
|
|
const GstRTPChannelOrder *order;
|
|
GstAudioInfo *info;
|
|
|
|
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
payload = 96;
|
|
gst_structure_get_int (structure, "payload", &payload);
|
|
switch (payload) {
|
|
case GST_RTP_PAYLOAD_L16_STEREO:
|
|
channels = 2;
|
|
clock_rate = 44100;
|
|
break;
|
|
case GST_RTP_PAYLOAD_L16_MONO:
|
|
channels = 1;
|
|
clock_rate = 44100;
|
|
break;
|
|
default:
|
|
/* no fixed mapping, we need clock-rate */
|
|
channels = 0;
|
|
clock_rate = 0;
|
|
break;
|
|
}
|
|
|
|
/* caps can overwrite defaults */
|
|
clock_rate =
|
|
gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
|
|
if (clock_rate == 0)
|
|
goto no_clockrate;
|
|
|
|
channels =
|
|
gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
|
|
if (channels == 0) {
|
|
channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
|
|
if (channels == 0) {
|
|
/* channels defaults to 1 otherwise */
|
|
channels = 1;
|
|
}
|
|
}
|
|
|
|
depayload->clock_rate = clock_rate;
|
|
|
|
info = &rtpL16depay->info;
|
|
gst_audio_info_init (info);
|
|
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
|
|
info->rate = clock_rate;
|
|
info->channels = channels;
|
|
info->bpf = (info->finfo->width / 8) * channels;
|
|
|
|
/* add channel positions */
|
|
channel_order = gst_structure_get_string (structure, "channel-order");
|
|
|
|
order = gst_rtp_channels_get_by_order (channels, channel_order);
|
|
rtpL16depay->order = order;
|
|
if (order) {
|
|
memcpy (info->position, order->pos,
|
|
sizeof (GstAudioChannelPosition) * channels);
|
|
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
|
|
} else {
|
|
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
|
|
(NULL), ("Unknown channel order '%s' for %d channels",
|
|
GST_STR_NULL (channel_order), channels));
|
|
/* create default NONE layout */
|
|
gst_rtp_channels_create_default (channels, info->position);
|
|
}
|
|
|
|
srccaps = gst_audio_info_to_caps (info);
|
|
res = gst_pad_set_caps (depayload->srcpad, srccaps);
|
|
gst_caps_unref (srccaps);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_clockrate:
|
|
{
|
|
GST_ERROR_OBJECT (depayload, "no clock-rate specified");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
|
|
{
|
|
GstRtpL16Depay *rtpL16depay;
|
|
GstBuffer *outbuf;
|
|
gint payload_len;
|
|
gboolean marker;
|
|
|
|
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
|
|
|
|
payload_len = gst_rtp_buffer_get_payload_len (rtp);
|
|
|
|
if (payload_len <= 0)
|
|
goto empty_packet;
|
|
|
|
GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
|
|
|
|
outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
|
|
marker = gst_rtp_buffer_get_marker (rtp);
|
|
|
|
if (marker) {
|
|
/* mark talk spurt with RESYNC */
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
|
|
}
|
|
|
|
outbuf = gst_buffer_make_writable (outbuf);
|
|
if (rtpL16depay->order &&
|
|
!gst_audio_buffer_reorder_channels (outbuf,
|
|
rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
|
|
rtpL16depay->info.position, rtpL16depay->order->pos)) {
|
|
goto reorder_failed;
|
|
}
|
|
|
|
gst_rtp_drop_non_audio_meta (rtpL16depay, outbuf);
|
|
|
|
return outbuf;
|
|
|
|
/* ERRORS */
|
|
empty_packet:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
|
|
("Empty Payload."), (NULL));
|
|
return NULL;
|
|
}
|
|
reorder_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
|
|
("Channel reordering failed."), (NULL));
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpL16depay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY);
|
|
}
|