gstreamer/gst/rtp/gstrtpg729pay.c
2021-03-29 12:45:22 +02:00

394 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2007> Nokia Corporation
* Copyright (C) <2007> Collabora Ltd
* @author: Olivier Crete <olivier.crete@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.
*/
/*
* This payloader assumes that the data will ALWAYS come as zero or more
* 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
* Any other buffer format won't work
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/base/gstadapter.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtpg729pay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpg729pay_debug);
#define GST_CAT_DEFAULT (rtpg729pay_debug)
#define G729_FRAME_SIZE 10
#define G729B_CN_FRAME_SIZE 2
#define G729_FRAME_DURATION (10 * GST_MSECOND)
#define G729_FRAME_DURATION_MS (10)
static gboolean
gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps);
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf);
static GstStateChangeReturn
gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition);
static GstStaticPadTemplate gst_rtp_g729_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/G729, " /* according to RFC 3555 */
"channels = (int) 1, " "rate = (int) 8000")
);
static GstStaticPadTemplate gst_rtp_g729_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_G729_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"G729\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"G729\"")
);
#define gst_rtp_g729_pay_parent_class parent_class
G_DEFINE_TYPE (GstRTPG729Pay, gst_rtp_g729_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpg729pay, "rtpg729pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G729_PAY, rtp_element_init (plugin));
static void
gst_rtp_g729_pay_finalize (GObject * object)
{
GstRTPG729Pay *pay = GST_RTP_G729_PAY (object);
g_object_unref (pay->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
"G.729 RTP Payloader");
gobject_class->finalize = gst_rtp_g729_pay_finalize;
gstelement_class->change_state = gst_rtp_g729_pay_change_state;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g729_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g729_pay_src_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP G.729 payloader", "Codec/Payloader/Network/RTP",
"Packetize G.729 audio into RTP packets",
"Olivier Crete <olivier.crete@collabora.co.uk>");
payload_class->set_caps = gst_rtp_g729_pay_set_caps;
payload_class->handle_buffer = gst_rtp_g729_pay_handle_buffer;
}
static void
gst_rtp_g729_pay_init (GstRTPG729Pay * pay)
{
GstRTPBasePayload *payload = GST_RTP_BASE_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
pay->adapter = gst_adapter_new ();
}
static void
gst_rtp_g729_pay_reset (GstRTPG729Pay * pay)
{
gst_adapter_clear (pay->adapter);
pay->discont = FALSE;
pay->next_rtp_time = 0;
pay->first_ts = GST_CLOCK_TIME_NONE;
pay->first_rtp_time = 0;
}
static gboolean
gst_rtp_g729_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_G729, "G729", 8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
gst_rtp_g729_pay_push (GstRTPG729Pay * rtpg729pay, GstBuffer * buf)
{
GstRTPBasePayload *basepayload;
GstClockTime duration;
guint frames;
GstBuffer *outbuf;
GstFlowReturn ret;
GstRTPBuffer rtp = { NULL };
guint payload_len = gst_buffer_get_size (buf);
basepayload = GST_RTP_BASE_PAYLOAD (rtpg729pay);
GST_DEBUG_OBJECT (rtpg729pay, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (rtpg729pay->next_ts));
/* create buffer to hold the payload */
outbuf =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(rtpg729pay), 0, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_READWRITE, &rtp);
/* set metadata */
frames =
(payload_len / G729_FRAME_SIZE) + ((payload_len % G729_FRAME_SIZE) >> 1);
duration = frames * G729_FRAME_DURATION;
GST_BUFFER_PTS (outbuf) = rtpg729pay->next_ts;
GST_BUFFER_DURATION (outbuf) = duration;
GST_BUFFER_OFFSET (outbuf) = rtpg729pay->next_rtp_time;
rtpg729pay->next_ts += duration;
rtpg729pay->next_rtp_time += frames * 80;
if (G_UNLIKELY (rtpg729pay->discont)) {
GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_set_marker (&rtp, TRUE);
rtpg729pay->discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
/* append payload */
gst_rtp_copy_audio_meta (basepayload, outbuf, buf);
outbuf = gst_buffer_append (outbuf, buf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
}
static void
gst_rtp_g729_pay_recalc_rtp_time (GstRTPG729Pay * rtpg729pay, GstClockTime time)
{
if (GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts)
&& GST_CLOCK_TIME_IS_VALID (time) && time >= rtpg729pay->first_ts) {
GstClockTime diff;
guint32 rtpdiff;
diff = time - rtpg729pay->first_ts;
rtpdiff = (diff / GST_MSECOND) * 8;
rtpg729pay->next_rtp_time = rtpg729pay->first_rtp_time + rtpdiff;
GST_DEBUG_OBJECT (rtpg729pay,
"elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
"new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
rtpg729pay->next_rtp_time);
}
}
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstRTPG729Pay *rtpg729pay = GST_RTP_G729_PAY (payload);
GstAdapter *adapter = NULL;
guint payload_len;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gsize size;
GstClockTime timestamp;
size = gst_buffer_get_size (buf);
if (size % G729_FRAME_SIZE != 0 &&
size % G729_FRAME_SIZE != G729B_CN_FRAME_SIZE)
goto invalid_size;
/* max number of bytes based on given ptime, has to be multiple of
* frame_duration */
if (payload->max_ptime != -1) {
guint ptime_ms = payload->max_ptime / GST_MSECOND;
maxptime_octets = G729_FRAME_SIZE *
(int) (ptime_ms / G729_FRAME_DURATION_MS);
if (maxptime_octets < G729_FRAME_SIZE) {
GST_WARNING_OBJECT (payload, "Given ptime %" G_GINT64_FORMAT
" is smaller than minimum %d ns, overwriting to minimum",
payload->max_ptime, G729_FRAME_DURATION_MS);
maxptime_octets = G729_FRAME_SIZE;
}
}
max_payload_len = MIN (
/* MTU max */
(int) (gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU
(payload), 0, 0) / G729_FRAME_SIZE)
* G729_FRAME_SIZE,
/* ptime max */
maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple
of frame duration */
{
guint64 min_ptime = payload->min_ptime;
min_ptime = min_ptime / GST_MSECOND;
minptime_octets = G729_FRAME_SIZE *
(int) (min_ptime / G729_FRAME_DURATION_MS);
}
min_payload_len = MAX (minptime_octets, G729_FRAME_SIZE);
if (min_payload_len > max_payload_len) {
min_payload_len = max_payload_len;
}
/* If the ptime is specified in the caps, tried to adhere to it exactly */
if (payload->ptime) {
guint64 ptime = payload->ptime / GST_MSECOND;
guint ptime_in_bytes = G729_FRAME_SIZE *
(guint) (ptime / G729_FRAME_DURATION_MS);
/* clip to computed min and max lengths */
ptime_in_bytes = MAX (min_payload_len, ptime_in_bytes);
ptime_in_bytes = MIN (max_payload_len, ptime_in_bytes);
min_payload_len = max_payload_len = ptime_in_bytes;
}
GST_LOG_OBJECT (payload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
adapter = rtpg729pay->adapter;
available = gst_adapter_available (adapter);
timestamp = GST_BUFFER_PTS (buf);
/* resync rtp time on discont or a discontinuous cn packet */
if (GST_BUFFER_IS_DISCONT (buf)) {
/* flush remainder */
if (available > 0) {
gst_rtp_g729_pay_push (rtpg729pay,
gst_adapter_take_buffer_fast (adapter, available));
available = 0;
}
rtpg729pay->discont = TRUE;
gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
}
if (size < G729_FRAME_SIZE)
gst_rtp_g729_pay_recalc_rtp_time (rtpg729pay, timestamp);
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpg729pay->first_ts))) {
rtpg729pay->first_ts = timestamp;
rtpg729pay->first_rtp_time = rtpg729pay->next_rtp_time;
}
/* let's reset the base timestamp when the adapter is empty */
if (available == 0)
rtpg729pay->next_ts = timestamp;
if (available == 0 && size >= min_payload_len && size <= max_payload_len) {
ret = gst_rtp_g729_pay_push (rtpg729pay, buf);
return ret;
}
gst_adapter_push (adapter, buf);
available = gst_adapter_available (adapter);
/* as long as we have full frames */
/* this loop will push all available buffers till the last frame */
while (available >= min_payload_len ||
available % G729_FRAME_SIZE == G729B_CN_FRAME_SIZE) {
/* We send as much as we can */
if (available <= max_payload_len) {
payload_len = available;
} else {
payload_len = MIN (max_payload_len,
(available / G729_FRAME_SIZE) * G729_FRAME_SIZE);
}
ret = gst_rtp_g729_pay_push (rtpg729pay,
gst_adapter_take_buffer_fast (adapter, payload_len));
available -= payload_len;
}
return ret;
/* ERRORS */
invalid_size:
{
GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
("Invalid input buffer size"),
("Invalid buffer size, should be a multiple of"
" G729_FRAME_SIZE(10) with an optional G729B_CN_FRAME_SIZE(2)"
" added to it, but it is %" G_GSIZE_FORMAT, size));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_rtp_g729_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
/* handle upwards state changes here */
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
/* handle downwards state changes */
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_g729_pay_reset (GST_RTP_G729_PAY (element));
break;
default:
break;
}
return ret;
}