gstreamer/gst/rtp/gstrtpg729pay.c
Wim Taymans 5e27695ca2 gst/rtp/: Fix the descriptions and fix some email addresses.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpL16pay.h:
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps):
* gst/rtp/gstrtpac3depay.h:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpamrpay.h:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpdepay.h:
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps):
* gst/rtp/gstrtpg726depay.c:
* gst/rtp/gstrtpg726pay.c:
* gst/rtp/gstrtpg729depay.c:
* gst/rtp/gstrtpg729pay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps):
* gst/rtp/gstrtph263depay.h:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pay.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
* gst/rtp/gstrtph263pdepay.h:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph263ppay.h:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtph264depay.h:
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtph264pay.h:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpjpegdepay.h:
* gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps):
* gst/rtp/gstrtpmp1sdepay.h:
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
* gst/rtp/gstrtpmp2tdepay.h:
* gst/rtp/gstrtpmp2tpay.c:
* gst/rtp/gstrtpmp2tpay.h:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps):
* gst/rtp/gstrtpmp4apay.c:
* gst/rtp/gstrtpmp4apay.h:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps):
* gst/rtp/gstrtpmp4gdepay.h:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
* gst/rtp/gstrtpmp4vdepay.h:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
* gst/rtp/gstrtpmp4vpay.h:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpadepay.h:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtpmpapay.h:
* gst/rtp/gstrtpmpvdepay.c:
* gst/rtp/gstrtpmpvdepay.h:
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtpsv3vdepay.h:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheoradepay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtptheorapay.h:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
* gst/rtp/gstrtpvorbispay.h:
* gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
* gst/rtp/gstrtpvrawpay.c:
Fix the descriptions and fix some email addresses.
2008-11-25 18:03:02 +00:00

307 lines
9.3 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"
/* TODO: fix gstrtpbuffer.h */
#undef GST_RTP_PAYLOAD_G729
#define GST_RTP_PAYLOAD_G729 18
#undef GST_RTP_PAYLOAD_G729_STRING
#define GST_RTP_PAYLOAD_G729_STRING "18"
#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\"")
);
static void
gst_rtp_g729_pay_init (GstRTPG729Pay * pay, GstRTPG729PayClass * klass);
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);
}
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;
const guint8 *data = NULL;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gboolean use_adapter = FALSE;
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 / 1000000;
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;
g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
min_ptime = min_ptime / 1000000;
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;
}
GST_DEBUG_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);
if (adapter && gst_adapter_available (adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (adapter, buf);
available = gst_adapter_available (adapter);
use_adapter = TRUE;
} else {
/* let's set the base timestamp */
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
/* If buffer fits on an RTP packet, let's just push it through */
/* this will check against max_ptime and max_mtu */
if (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);
return ret;
}
available = GST_BUFFER_SIZE (buf);
data = (guint8 *) GST_BUFFER_DATA (buf);
}
/* 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);
}
if (use_adapter) {
data = gst_adapter_peek (adapter, payload_len);
}
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
payload_len, basertpaudiopayload->base_ts);
num = payload_len / G729_FRAME_SIZE;
basertpaudiopayload->base_ts += G729_FRAME_DURATION * num;
if (use_adapter) {
gst_adapter_flush (adapter, payload_len);
available = gst_adapter_available (adapter);
} else {
available -= payload_len;
data += payload_len;
}
}
if (!use_adapter) {
if (available != 0 && adapter) {
GstBuffer *buf2;
buf2 = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - available, available);
gst_adapter_push (adapter, buf2);
} else {
gst_buffer_unref (buf);
}
}
if (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);
}