gstreamer/gst/rtp/gstrtpsbcdepay.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

288 lines
8.3 KiB
C

/*
* 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_static_pad_template (element_class,
&gst_rtp_sbc_depay_src_template);
gst_element_class_add_static_pad_template (element_class,
&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);
if (oldcaps)
gst_caps_unref (oldcaps);
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_non_audio_meta (depay, data);
} 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);
}