gstreamer/gst/rtp/gstrtpsbcdepay.c

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/*
* GStreamer RTP SBC depayloader
*
* Copyright (C) 2012 Collabora Ltd.
* @author: Arun Raghavan <arun.raghavan@collabora.co.uk>
*
* 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 "gstrtpsbcdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
#define GST_CAT_DEFAULT (rtpsbcdepay_debug)
static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-sbc, "
"rate = (int) { 16000, 32000, 44100, 48000 }, "
"channels = (int) [ 1, 2 ], "
"mode = (string) { mono, dual, stereo, joint }, "
"blocks = (int) { 4, 8, 12, 16 }, "
"subbands = (int) { 4, 8 }, "
"allocation-method = (string) { snr, loudness }, "
"bitpool = (int) [ 2, 64 ]")
);
static GstStaticPadTemplate gst_rtp_sbc_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) audio,"
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
"encoding-name = (string) SBC")
);
#define gst_rtp_sbc_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
static void gst_rtp_sbc_depay_finalize (GObject * object);
static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
GstCaps * caps);
static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
GstRTPBuffer * rtp);
static void
gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
{
GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtp_sbc_depay_finalize;
gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
gstbasertpdepayload_class->process_rtp_packet = gst_rtp_sbc_depay_process;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_depay_sink_template));
GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
"SBC Audio RTP Depayloader");
gst_element_class_set_static_metadata (element_class,
"RTP SBC audio depayloader",
"Codec/Depayloader/Network/RTP",
"Extracts SBC audio from RTP packets",
"Arun Raghavan <arun.raghavan@collabora.co.uk>");
}
static void
gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
{
rtpsbcdepay->adapter = gst_adapter_new ();
}
static void
gst_rtp_sbc_depay_finalize (GObject * object)
{
GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
gst_object_unref (depay->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
* simple way to consolidate the two. This is best done by moving the function
* to the codec-utils library in gst-plugins-base when these elements move to
* GStreamer. */
static int
gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
gint size, int *framelen, int *samples)
{
int blocks, channel_mode, channels, subbands, bitpool;
int length;
if (size < 3) {
/* Not enough data for the header */
return -1;
}
/* Sanity check */
if (data[0] != 0x9c) {
GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
return -2;
}
blocks = (data[1] >> 4) & 0x3;
blocks = (blocks + 1) * 4;
channel_mode = (data[1] >> 2) & 0x3;
channels = channel_mode ? 2 : 1;
subbands = (data[1] & 0x1);
subbands = (subbands + 1) * 4;
bitpool = data[2];
length = 4 + ((4 * subbands * channels) / 8);
if (channel_mode == 0 || channel_mode == 1) {
/* Mono || Dual channel */
length += ((blocks * channels * bitpool)
+ 4 /* round up */ ) / 8;
} else {
/* Stereo || Joint stereo */
gboolean joint = (channel_mode == 3);
length += ((joint * subbands) + (blocks * bitpool)
+ 4 /* round up */ ) / 8;
}
*framelen = length;
*samples = blocks * subbands;
return 0;
}
static gboolean
gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
{
GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
GstStructure *structure;
GstCaps *outcaps, *oldcaps;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
goto bad_caps;
outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
depay->rate, NULL);
gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
/* Caps have changed, flush old data */
gst_adapter_clear (depay->adapter);
}
gst_caps_unref (outcaps);
return TRUE;
bad_caps:
GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
GST_PTR_FORMAT, caps);
return FALSE;
}
static GstBuffer *
gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstRTPBuffer * rtp)
{
GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
GstBuffer *data = NULL;
gboolean fragment, start, last;
guint8 nframes;
guint8 *payload;
guint payload_len;
GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
gst_buffer_get_size (rtp->buffer));
if (gst_rtp_buffer_get_marker (rtp)) {
/* Marker isn't supposed to be set */
GST_WARNING_OBJECT (depay, "Marker bit was set");
goto bad_packet;
}
payload = gst_rtp_buffer_get_payload (rtp);
payload_len = gst_rtp_buffer_get_payload_len (rtp);
fragment = payload[0] & 0x80;
start = payload[0] & 0x40;
last = payload[0] & 0x20;
nframes = payload[0] & 0x0f;
payload += 1;
payload_len -= 1;
data = gst_rtp_buffer_get_payload_subbuffer (rtp, 1, -1);
if (fragment) {
/* Got a packet with a fragment */
GST_LOG_OBJECT (depay, "Got fragment");
if (start && gst_adapter_available (depay->adapter)) {
GST_WARNING_OBJECT (depay, "Missing last fragment");
gst_adapter_clear (depay->adapter);
} else if (!start && !gst_adapter_available (depay->adapter)) {
GST_WARNING_OBJECT (depay, "Missing start fragment");
gst_buffer_unref (data);
data = NULL;
goto out;
}
gst_adapter_push (depay->adapter, data);
if (last) {
data = gst_adapter_take_buffer (depay->adapter,
gst_adapter_available (depay->adapter));
gst_rtp_drop_meta (GST_ELEMENT_CAST (depay), data,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
} else
data = NULL;
} else {
/* !fragment */
gint framelen, samples;
GST_LOG_OBJECT (depay, "Got %d frames", nframes);
if (gst_rtp_sbc_depay_get_params (depay, payload,
payload_len, &framelen, &samples) < 0) {
gst_adapter_clear (depay->adapter);
goto bad_packet;
}
GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
if (nframes * framelen > (gint) payload_len) {
GST_WARNING_OBJECT (depay, "Short packet");
goto bad_packet;
} else if (nframes * framelen < (gint) payload_len) {
GST_WARNING_OBJECT (depay, "Junk at end of packet");
}
}
out:
return data;
bad_packet:
GST_ELEMENT_WARNING (depay, STREAM, DECODE,
("Received invalid RTP payload, dropping"), (NULL));
goto out;
}
gboolean
gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
GST_TYPE_RTP_SBC_DEPAY);
}