gstreamer/gst/rtp/gstrtpceltpay.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

502 lines
14 KiB
C

/* 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpceltpay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpceltpay_debug);
#define GST_CAT_DEFAULT (rtpceltpay_debug)
static GstStaticPadTemplate gst_rtp_celt_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-celt, "
"rate = (int) [ 32000, 64000 ], "
"channels = (int) [1, 2], " "frame-size = (int) [ 64, 512 ]")
);
static GstStaticPadTemplate gst_rtp_celt_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [ 32000, 48000 ], "
"encoding-name = (string) \"CELT\"")
);
static void gst_rtp_celt_pay_finalize (GObject * object);
static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstCaps *gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
#define gst_rtp_celt_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpCELTPay, gst_rtp_celt_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0,
"CELT RTP Payloader");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_celt_pay_finalize;
gstelement_class->change_state = gst_rtp_celt_pay_change_state;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_celt_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_celt_pay_src_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP CELT payloader",
"Codec/Payloader/Network/RTP",
"Payload-encodes CELT audio into a RTP packet",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_celt_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_celt_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_celt_pay_handle_buffer;
}
static void
gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay)
{
rtpceltpay->queue = g_queue_new ();
}
static void
gst_rtp_celt_pay_finalize (GObject * object)
{
GstRtpCELTPay *rtpceltpay;
rtpceltpay = GST_RTP_CELT_PAY (object);
g_queue_free (rtpceltpay->queue);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_celt_pay_clear_queued (GstRtpCELTPay * rtpceltpay)
{
GstBuffer *buf;
while ((buf = g_queue_pop_head (rtpceltpay->queue)))
gst_buffer_unref (buf);
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
}
static void
gst_rtp_celt_pay_add_queued (GstRtpCELTPay * rtpceltpay, GstBuffer * buffer,
guint ssize, guint size, GstClockTime duration)
{
g_queue_push_tail (rtpceltpay->queue, buffer);
rtpceltpay->sbytes += ssize;
rtpceltpay->bytes += size;
/* only add durations when we have a valid previous duration */
if (rtpceltpay->qduration != -1) {
if (duration != -1)
/* only add valid durations */
rtpceltpay->qduration += duration;
else
/* if we add a buffer without valid duration, our total queued duration
* becomes unknown */
rtpceltpay->qduration = -1;
}
}
static gboolean
gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
/* don't configure yet, we wait for the ident packet */
return TRUE;
}
static GstCaps *
gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
GstCaps * filter)
{
GstCaps *otherpadcaps;
GstCaps *caps;
const gchar *params;
caps = gst_pad_get_pad_template_caps (pad);
otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *ps;
GstStructure *s;
gint clock_rate = 0, frame_size = 0, channels = 1;
caps = gst_caps_make_writable (caps);
ps = gst_caps_get_structure (otherpadcaps, 0);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
}
if ((params = gst_structure_get_string (ps, "frame-size")))
frame_size = atoi (params);
if (frame_size)
gst_structure_set (s, "frame-size", G_TYPE_INT, frame_size, NULL);
if ((params = gst_structure_get_string (ps, "encoding-params"))) {
channels = atoi (params);
gst_structure_fixate_field_nearest_int (s, "channels", channels);
}
GST_DEBUG_OBJECT (payload, "clock-rate=%d frame-size=%d channels=%d",
clock_rate, frame_size, channels);
}
gst_caps_unref (otherpadcaps);
}
if (filter) {
GstCaps *tmp;
GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"
GST_PTR_FORMAT, caps, filter);
tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
return caps;
}
static gboolean
gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,
const guint8 * data, guint size)
{
guint32 version, header_size, rate, nb_channels, frame_size, overlap;
guint32 bytes_per_packet;
GstRTPBasePayload *payload;
gchar *cstr, *fsstr;
gboolean res;
/* we need the header string (8), the version string (20), the version
* and the header length. */
if (size < 36)
goto too_small;
if (!g_str_has_prefix ((const gchar *) data, "CELT "))
goto wrong_header;
/* skip header and version string */
data += 28;
version = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version);
#if 0
if (version != 1)
goto wrong_version;
#endif
data += 4;
/* ensure sizes */
header_size = GST_READ_UINT32_LE (data);
if (header_size < 56)
goto header_too_small;
if (size < header_size)
goto payload_too_small;
data += 4;
rate = GST_READ_UINT32_LE (data);
data += 4;
nb_channels = GST_READ_UINT32_LE (data);
data += 4;
frame_size = GST_READ_UINT32_LE (data);
data += 4;
overlap = GST_READ_UINT32_LE (data);
data += 4;
bytes_per_packet = GST_READ_UINT32_LE (data);
GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d",
rate, nb_channels, frame_size);
GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d",
overlap, bytes_per_packet);
payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);
gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);
cstr = g_strdup_printf ("%d", nb_channels);
fsstr = g_strdup_printf ("%d", frame_size);
res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);
g_free (cstr);
g_free (fsstr);
return res;
/* ERRORS */
too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet too small, need at least 32 bytes");
return FALSE;
}
wrong_header:
{
GST_DEBUG_OBJECT (rtpceltpay,
"ident packet does not start with \"CELT \"");
return FALSE;
}
#if 0
wrong_version:
{
GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d",
version);
return FALSE;
}
#endif
header_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"header size too small, need at least 80 bytes, " "got only %d",
header_size);
return FALSE;
}
payload_too_small:
{
GST_DEBUG_OBJECT (rtpceltpay,
"payload too small, need at least %d bytes, got only %d", header_size,
size);
return FALSE;
}
}
static GstFlowReturn
gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)
{
GstFlowReturn ret;
GstBuffer *buf, *outbuf;
guint8 *payload, *spayload;
guint payload_len;
GstClockTime duration;
GstRTPBuffer rtp = { NULL, };
payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;
duration = rtpceltpay->qduration;
GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpceltpay->qduration));
/* get a big enough packet for the sizes + payloads */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
GST_BUFFER_DURATION (outbuf) = duration;
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* point to the payload for size headers and data */
spayload = gst_rtp_buffer_get_payload (&rtp);
payload = spayload + rtpceltpay->sbytes;
while ((buf = g_queue_pop_head (rtpceltpay->queue))) {
guint size;
/* copy first timestamp to output */
if (GST_BUFFER_PTS (outbuf) == -1)
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf);
/* write the size to the header */
size = gst_buffer_get_size (buf);
while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
/* copy payload */
size = gst_buffer_get_size (buf);
gst_buffer_extract (buf, 0, payload, size);
payload += size;
gst_rtp_copy_audio_meta (rtpceltpay, outbuf, buf);
gst_buffer_unref (buf);
}
gst_rtp_buffer_unmap (&rtp);
/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
return ret;
}
static GstFlowReturn
gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstRtpCELTPay *rtpceltpay;
gsize payload_len;
GstMapInfo map;
GstClockTime duration, packet_dur;
guint i, ssize, packet_len;
rtpceltpay = GST_RTP_CELT_PAY (basepayload);
ret = GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
switch (rtpceltpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))
goto parse_error;
goto cleanup;
case 1:
/* comment packet, we ignore it */
goto cleanup;
default:
/* other packets go in the payload */
break;
}
gst_buffer_unmap (buffer, &map);
duration = GST_BUFFER_DURATION (buffer);
GST_LOG_OBJECT (rtpceltpay,
"got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,
GST_TIME_ARGS (duration), map.size);
/* calculate the size of the size field and the payload */
ssize = 1;
for (i = map.size; i > 0xff; i -= 0xff)
ssize++;
GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);
/* calculate what the new size and duration would be of the packet */
payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;
if (rtpceltpay->qduration != -1 && duration != -1)
packet_dur = rtpceltpay->qduration + duration;
else
packet_dur = 0;
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {
/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
}
/* queue the packet */
gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);
done:
rtpceltpay->packet++;
return ret;
/* ERRORS */
cleanup:
{
gst_buffer_unmap (buffer, &map);
goto done;
}
parse_error:
{
GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
gst_buffer_unmap (buffer, &map);
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpCELTPay *rtpceltpay;
GstStateChangeReturn ret;
rtpceltpay = GST_RTP_CELT_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtpceltpay->packet = 0;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_celt_pay_clear_queued (rtpceltpay);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_celt_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpceltpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY);
}