/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2005 Edgard Lima <edgard.lima@gmail.com>
* Copyright (C) 2005 Nokia Corporation <kai.vehmanen@nokia.com>
* Copyright (C) 2007,2008 Axis Communications <dev-gstreamer@axis.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 "gstrtpg726pay.h"
GST_DEBUG_CATEGORY_STATIC (rtpg726pay_debug);
#define GST_CAT_DEFAULT (rtpg726pay_debug)
#define DEFAULT_FORCE_AAL2 TRUE
enum
{
PROP_0,
PROP_FORCE_AAL2
};
static GstStaticPadTemplate gst_rtp_g726_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
"channels = (int) 1, "
"rate = (int) 8000, "
"bitrate = (int) { 16000, 24000, 32000, 40000 }, "
"layout = (string) \"g726\"")
);
static GstStaticPadTemplate gst_rtp_g726_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) 8000, "
"encoding-name = (string) { \"G726-16\", \"G726-24\", \"G726-32\", \"G726-40\", "
" \"AAL2-G726-16\", \"AAL2-G726-24\", \"AAL2-G726-32\", \"AAL2-G726-40\" } ")
);
static void gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
#define gst_rtp_g726_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpG726Pay, gst_rtp_g726_pay,
GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
static void
gst_rtp_g726_pay_class_init (GstRtpG726PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->set_property = gst_rtp_g726_pay_set_property;
gobject_class->get_property = gst_rtp_g726_pay_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_FORCE_AAL2,
g_param_spec_boolean ("force-aal2", "Force AAL2",
"Force AAL2 encoding for compatibility with bad depayloaders",
DEFAULT_FORCE_AAL2, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g726_pay_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_g726_pay_src_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP G.726 payloader", "Codec/Payloader/Network/RTP",
"Payload-encodes G.726 audio into a RTP packet",
"Axis Communications <dev-gstreamer@axis.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_g726_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_g726_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpg726pay_debug, "rtpg726pay", 0,
"G.726 RTP Payloader");
}
static void
gst_rtp_g726_pay_init (GstRtpG726Pay * rtpg726pay)
{
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (rtpg726pay);
GST_RTP_BASE_PAYLOAD (rtpg726pay)->clock_rate = 8000;
rtpg726pay->force_aal2 = DEFAULT_FORCE_AAL2;
/* sample based codec */
gst_rtp_base_audio_payload_set_sample_based (rtpbaseaudiopayload);
}
static gboolean
gst_rtp_g726_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gchar *encoding_name;
GstStructure *structure;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
GstRtpG726Pay *pay;
GstCaps *peercaps;
gboolean res;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (payload);
pay = GST_RTP_G726_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "bitrate", &pay->bitrate))
pay->bitrate = 32000;
GST_DEBUG_OBJECT (payload, "using bitrate %d", pay->bitrate);
pay->aal2 = FALSE;
/* first see what we can do with the bitrate */
switch (pay->bitrate) {
case 16000:
encoding_name = g_strdup ("G726-16");
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
2);
break;
case 24000:
encoding_name = g_strdup ("G726-24");
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
3);
break;
case 32000:
encoding_name = g_strdup ("G726-32");
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
4);
break;
case 40000:
encoding_name = g_strdup ("G726-40");
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
5);
break;
default:
goto invalid_bitrate;
}
GST_DEBUG_OBJECT (payload, "selected base encoding %s", encoding_name);
/* now see if we need to produce AAL2 or not */
peercaps = gst_pad_peer_query_caps (payload->srcpad, NULL);
if (peercaps) {
GstCaps *filter, *intersect;
gchar *capsstr;
GST_DEBUG_OBJECT (payload, "have peercaps %" GST_PTR_FORMAT, peercaps);
capsstr = g_strdup_printf ("application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 8000, "
"encoding-name = (string) %s; "
"application/x-rtp, "
"media = (string) \"audio\", "
"clock-rate = (int) 8000, "
"encoding-name = (string) AAL2-%s", encoding_name, encoding_name);
filter = gst_caps_from_string (capsstr);
g_free (capsstr);
g_free (encoding_name);
/* intersect to filter */
intersect = gst_caps_intersect (peercaps, filter);
gst_caps_unref (peercaps);
gst_caps_unref (filter);
GST_DEBUG_OBJECT (payload, "intersected to %" GST_PTR_FORMAT, intersect);
if (!intersect)
goto no_format;
if (gst_caps_is_empty (intersect)) {
gst_caps_unref (intersect);
goto no_format;
}
structure = gst_caps_get_structure (intersect, 0);
/* now see what encoding name we settled on, we need to dup because the
* string goes away when we unref the intersection below. */
encoding_name =
g_strdup (gst_structure_get_string (structure, "encoding-name"));
/* if we managed to negotiate to AAL2, we definatly are going to do AAL2
* encoding. Else we only encode AAL2 when explicitly set by the
* property. */
if (g_str_has_prefix (encoding_name, "AAL2-"))
pay->aal2 = TRUE;
else
pay->aal2 = pay->force_aal2;
GST_DEBUG_OBJECT (payload, "final encoding %s, AAL2 %d", encoding_name,
pay->aal2);
gst_caps_unref (intersect);
} else {
/* downstream can do anything but we prefer the better supported non-AAL2 */
pay->aal2 = pay->force_aal2;
GST_DEBUG_OBJECT (payload, "no peer caps, AAL2 %d", pay->aal2);
}
gst_rtp_base_payload_set_options (payload, "audio", TRUE, encoding_name,
8000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
g_free (encoding_name);
return res;
/* ERRORS */
invalid_bitrate:
{
GST_ERROR_OBJECT (payload, "invalid bitrate %d specified", pay->bitrate);
return FALSE;
}
no_format:
{
GST_ERROR_OBJECT (payload, "could not negotiate format");
return FALSE;
}
}
static GstFlowReturn
gst_rtp_g726_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstFlowReturn res;
GstRtpG726Pay *pay;
pay = GST_RTP_G726_PAY (payload);
if (!pay->aal2) {
GstMapInfo map;
guint8 *data, tmp;
gsize size;
/* for non AAL2, we need to reshuffle the bytes, we can do this in-place
* when the buffer is writable. */
buffer = gst_buffer_make_writable (buffer);
gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
data = map.data;
size = map.size;
GST_LOG_OBJECT (pay, "packing %" G_GSIZE_FORMAT " bytes of data", map.size);
/* we need to reshuffle the bytes, output is of the form:
* A B C D .. with the number of bits depending on the bitrate. */
switch (pay->bitrate) {
case 16000:
{
/* 0
* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+-
* |D D|C C|B B|A A| ...
* |0 1|0 1|0 1|0 1|
* +-+-+-+-+-+-+-+-+-
*/
while (size > 0) {
tmp = *data;
*data++ = ((tmp & 0xc0) >> 6) |
((tmp & 0x30) >> 2) | ((tmp & 0x0c) << 2) | ((tmp & 0x03) << 6);
size--;
}
break;
}
case 24000:
{
/* 0 1 2
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
* |C C|B B B|A A A|F|E E E|D D D|C|H H H|G G G|F F| ...
* |1 2|0 1 2|0 1 2|2|0 1 2|0 1 2|0|0 1 2|0 1 2|0 1|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
*/
while (size > 2) {
tmp = *data;
*data++ = ((tmp & 0xc0) >> 6) |
((tmp & 0x38) >> 1) | ((tmp & 0x07) << 5);
tmp = *data;
*data++ = ((tmp & 0x80) >> 7) |
((tmp & 0x70) >> 3) | ((tmp & 0x0e) << 4) | ((tmp & 0x01) << 7);
tmp = *data;
*data++ = ((tmp & 0xe0) >> 5) |
((tmp & 0x1c) >> 2) | ((tmp & 0x03) << 6);
size -= 3;
}
break;
}
case 32000:
{
/* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
* |B B B B|A A A A|D D D D|C C C C| ...
* |0 1 2 3|0 1 2 3|0 1 2 3|0 1 2 3|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
*/
while (size > 0) {
tmp = *data;
*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
size--;
}
break;
}
case 40000:
{
/* 0 1 2 3 4
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
* |B B B|A A A A A|D|C C C C C|B B|E E E E|D D D D|G G|F F F F F|E|H H H H H|G G G|
* |2 3 4|0 1 2 3 4|4|0 1 2 3 4|0 1|1 2 3 4|0 1 2 3|3 4|0 1 2 3 4|0|0 1 2 3 4|0 1 2|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
*/
while (size > 4) {
tmp = *data;
*data++ = ((tmp & 0xe0) >> 5) | ((tmp & 0x1f) << 3);
tmp = *data;
*data++ = ((tmp & 0x80) >> 7) |
((tmp & 0x7c) >> 2) | ((tmp & 0x03) << 6);
tmp = *data;
*data++ = ((tmp & 0xf0) >> 4) | ((tmp & 0x0f) << 4);
tmp = *data;
*data++ = ((tmp & 0xc0) >> 6) |
((tmp & 0x3e) << 2) | ((tmp & 0x01) << 7);
tmp = *data;
*data++ = ((tmp & 0xf8) >> 3) | ((tmp & 0x07) << 5);
size -= 5;
}
break;
}
}
gst_buffer_unmap (buffer, &map);
}
res =
GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (payload,
buffer);
return res;
}
static void
gst_rtp_g726_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpG726Pay *rtpg726pay;
rtpg726pay = GST_RTP_G726_PAY (object);
switch (prop_id) {
case PROP_FORCE_AAL2:
rtpg726pay->force_aal2 = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_g726_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpG726Pay *rtpg726pay;
rtpg726pay = GST_RTP_G726_PAY (object);
switch (prop_id) {
case PROP_FORCE_AAL2:
g_value_set_boolean (value, rtpg726pay->force_aal2);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_g726_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpg726pay",
GST_RANK_SECONDARY, GST_TYPE_RTP_G726_PAY);
}