gstreamer/gst/rtp/gstrtpg729pay.c
Benjamin Otte 1055aaa9cb Add -Wredundant-decls warning flag
Also fix compile issues
2010-03-17 19:35:10 +01:00

284 lines
8.9 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, 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 "gstrtpg729pay.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 (GstBaseRTPPayload * payload, GstCaps * caps);
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
static const GstElementDetails gst_rtp_g729_pay_details =
GST_ELEMENT_DETAILS ("RTP G.729 payloader",
"Codec/Payloader/Network",
"Packetize G.729 audio into RTP packets",
"Olivier Crete <olivier.crete@collabora.co.uk>");
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\"")
);
GST_BOILERPLATE (GstRTPG729Pay, gst_rtp_g729_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_g729_pay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g729_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_g729_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_g729_pay_details);
GST_DEBUG_CATEGORY_INIT (rtpg729pay_debug, "rtpg729pay", 0,
"G.729 RTP Payloader");
}
static void
gst_rtp_g729_pay_class_init (GstRTPG729PayClass * klass)
{
GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
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, GstRTPG729PayClass * klass)
{
GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
payload->pt = GST_RTP_PAYLOAD_G729;
gst_basertppayload_set_options (payload, "audio", FALSE, "G729", 8000);
gst_base_rtp_audio_payload_set_frame_based (audiopayload);
gst_base_rtp_audio_payload_set_frame_options (audiopayload,
G729_FRAME_DURATION_MS, G729_FRAME_SIZE);
}
static gboolean
gst_rtp_g729_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gboolean res;
GstStructure *structure;
gint pt;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "payload", &pt))
pt = GST_RTP_PAYLOAD_G729;
payload->pt = pt;
payload->dynamic = pt != GST_RTP_PAYLOAD_G729;
res = gst_basertppayload_set_outcaps (payload, NULL);
return res;
}
static GstFlowReturn
gst_rtp_g729_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
GstBaseRTPAudioPayload *basertpaudiopayload =
GST_BASE_RTP_AUDIO_PAYLOAD (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;
available = GST_BUFFER_SIZE (buf);
if (available % G729_FRAME_SIZE != 0 &&
available % 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 (basertpaudiopayload, "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_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 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->abidata.ABI.ptime) {
guint64 ptime = payload->abidata.ABI.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 (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
/* let's reset the base timestamp when the adapter is empty */
if (gst_adapter_available (adapter) == 0)
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
if (gst_adapter_available (adapter) == 0 &&
GST_BUFFER_SIZE (buf) >= min_payload_len &&
GST_BUFFER_SIZE (buf) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
GST_BUFFER_TIMESTAMP (buf));
gst_buffer_unref (buf);
g_object_unref (adapter);
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) {
guint num;
/* 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_base_rtp_audio_payload_flush (basertpaudiopayload, payload_len,
basertpaudiopayload->base_ts);
num = payload_len / G729_FRAME_SIZE;
basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
available = gst_adapter_available (adapter);
}
g_object_unref (adapter);
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 %u", available));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
gboolean
gst_rtp_g729_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg729pay",
GST_RANK_NONE, GST_TYPE_RTP_G729_PAY);
}