gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpceltpay.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 <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
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#include "gstrtpelements.h"
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#include "gstrtpceltpay.h"
#include "gstrtputils.h"
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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 ]")
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);
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);
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static gboolean gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstCaps *gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload,
GstPad * pad, GstCaps * filter);
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static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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#define gst_rtp_celt_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpCELTPay, gst_rtp_celt_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpceltpay, "rtpceltpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY, rtp_element_init (plugin));
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static void
gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0,
"CELT RTP Payloader");
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gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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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);
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gst_element_class_set_static_metadata (gstelement_class, "RTP CELT payloader",
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"Codec/Payloader/Network/RTP",
"Payload-encodes CELT audio into a RTP packet",
"Wim Taymans <wim.taymans@gmail.com>");
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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;
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}
static void
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gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay)
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{
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;
}
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}
static gboolean
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gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
/* don't configure yet, we wait for the ident packet */
return TRUE;
}
static GstCaps *
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gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
GstCaps * filter)
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{
GstCaps *otherpadcaps;
GstCaps *caps;
const gchar *params;
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caps = gst_pad_get_pad_template_caps (pad);
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otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
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if (otherpadcaps) {
if (!gst_caps_is_empty (otherpadcaps)) {
GstStructure *ps;
GstStructure *s;
gint clock_rate = 0, frame_size = 0, channels = 1;
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caps = gst_caps_make_writable (caps);
ps = gst_caps_get_structure (otherpadcaps, 0);
s = gst_caps_get_structure (caps, 0);
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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);
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}
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;
}
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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;
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GstRTPBasePayload *payload;
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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);
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payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);
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gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);
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cstr = g_strdup_printf ("%d", nb_channels);
fsstr = g_strdup_printf ("%d", frame_size);
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res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
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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;
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GstRTPBuffer rtp = { NULL, };
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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_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(rtpceltpay), payload_len, 0, 0);
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GST_BUFFER_DURATION (outbuf) = duration;
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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/* point to the payload for size headers and data */
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spayload = gst_rtp_buffer_get_payload (&rtp);
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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);
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/* write the size to the header */
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size = gst_buffer_get_size (buf);
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while (size > 0xff) {
*spayload++ = 0xff;
size -= 0xff;
}
*spayload++ = size;
/* copy payload */
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size = gst_buffer_get_size (buf);
gst_buffer_extract (buf, 0, payload, size);
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payload += size;
gst_rtp_copy_audio_meta (rtpceltpay, outbuf, buf);
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gst_buffer_unref (buf);
}
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gst_rtp_buffer_unmap (&rtp);
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/* we consumed it all */
rtpceltpay->bytes = 0;
rtpceltpay->sbytes = 0;
rtpceltpay->qduration = 0;
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ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);
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return ret;
}
static GstFlowReturn
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gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
{
GstFlowReturn ret;
GstRtpCELTPay *rtpceltpay;
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gsize payload_len;
GstMapInfo map;
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GstClockTime duration, packet_dur;
guint i, ssize, packet_len;
rtpceltpay = GST_RTP_CELT_PAY (basepayload);
ret = GST_FLOW_OK;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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switch (rtpceltpay->packet) {
case 0:
/* ident packet. We need to parse the headers to construct the RTP
* properties. */
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if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))
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goto parse_error;
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goto cleanup;
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case 1:
/* comment packet, we ignore it */
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goto cleanup;
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default:
/* other packets go in the payload */
break;
}
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gst_buffer_unmap (buffer, &map);
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duration = GST_BUFFER_DURATION (buffer);
GST_LOG_OBJECT (rtpceltpay,
"got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,
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GST_TIME_ARGS (duration), map.size);
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/* calculate the size of the size field and the payload */
ssize = 1;
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for (i = map.size; i > 0xff; i -= 0xff)
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ssize++;
GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);
/* calculate what the new size and duration would be of the packet */
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payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;
if (rtpceltpay->qduration != -1 && duration != -1)
packet_dur = rtpceltpay->qduration + duration;
else
packet_dur = 0;
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packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
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if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {
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/* size or duration would overflow the packet, flush the queued data */
ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);
}
/* queue the packet */
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gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);
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done:
rtpceltpay->packet++;
return ret;
/* ERRORS */
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cleanup:
{
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gst_buffer_unmap (buffer, &map);
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goto done;
}
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parse_error:
{
GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),
("Error parsing first identification packet."));
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gst_buffer_unmap (buffer, &map);
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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;
}