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gst/rtp/: Added MP4A-LATM payloader to match the depayloader.

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Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
(gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
(gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
(gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
(gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
* gst/rtp/gstrtpmp4apay.h:
Added MP4A-LATM payloader to match the depayloader.
This commit is contained in:
Wim Taymans 2008-09-26 13:55:48 +00:00
parent 1dcf0755c5
commit c77bfaacb4
5 changed files with 550 additions and 1 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

13
ChangeLog
View file

@ -1,3 +1,16 @@
2008-09-26 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_get_type),
(gst_rtp_mp4a_pay_base_init), (gst_rtp_mp4a_pay_class_init),
(gst_rtp_mp4a_pay_init), (gst_rtp_mp4a_pay_finalize),
(gst_rtp_mp4a_pay_parse_audio_config), (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps), (gst_rtp_mp4a_pay_handle_buffer),
(gst_rtp_mp4a_pay_change_state), (gst_rtp_mp4a_pay_plugin_init):
* gst/rtp/gstrtpmp4apay.h:
Added MP4A-LATM payloader to match the depayloader.
2008-09-25 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/videomixer/videomixer.c: (gst_videomixer_fill_queues),

4
gst/rtp/Makefile.am
View file

@ -42,6 +42,7 @@ libgstrtp_la_SOURCES = \
gstrtpmp4gdepay.c \
gstrtpmp4gpay.c \
gstrtpmp4adepay.c \
gstrtpmp4apay.c \
gstrtpspeexdepay.c \
gstrtpspeexpay.c \
gstrtpsv3vdepay.c \
@ -103,7 +104,8 @@ noinst_HEADERS = \
gstrtpmp4vpay.h \
gstrtpmp4gdepay.h \
gstrtpmp4gpay.h \
gstrtpmp4adepay.h \
gstrtpmp4adepay.h \
gstrtpmp4apay.h \
gstrtpdepay.h \
gstasteriskh263.h \
gstrtpspeexdepay.h \

4
gst/rtp/gstrtp.c
View file

@ -58,6 +58,7 @@
#include "gstrtpmp4vdepay.h"
#include "gstrtpmp4vpay.h"
#include "gstrtpmp4adepay.h"
#include "gstrtpmp4apay.h"
#include "gstrtpmp4gdepay.h"
#include "gstrtpmp4gpay.h"
#include "gstrtpspeexpay.h"
@ -181,6 +182,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_mp4v_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_mp4a_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_mp4a_depay_plugin_init (plugin))
return FALSE;

467
gst/rtp/gstrtpmp4apay.c Normal file
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@ -0,0 +1,467 @@
/* GStreamer
* Copyright (C) <2008> 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp4apay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
#define GST_CAT_DEFAULT (rtpmp4apay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_mp4apay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload MPEG4 audio as RTP packets (RFC 3016)",
"Wim Taymans <wim.taymans@gmail.com>");
static GstStaticPadTemplate gst_rtp_mp4a_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg," "mpegversion=(int) 4")
);
static GstStaticPadTemplate gst_rtp_mp4a_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) [1, MAX ], "
"encoding-name = (string) \"MP4A-LATM\""
/* All optional parameters
*
* "cpresent = (string) \"0\""
* "config="
*/
)
);
static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass);
static void gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass);
static void gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay);
static void gst_rtp_mp4a_pay_finalize (GObject * object);
static gboolean gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstStateChangeReturn gst_rtp_mp4a_pay_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstBaseRTPPayloadClass *parent_class = NULL;
static GType
gst_rtp_mp4a_pay_get_type (void)
{
static GType rtpmp4apay_type = 0;
if (!rtpmp4apay_type) {
static const GTypeInfo rtpmp4apay_info = {
sizeof (GstRtpMP4APayClass),
(GBaseInitFunc) gst_rtp_mp4a_pay_base_init,
NULL,
(GClassInitFunc) gst_rtp_mp4a_pay_class_init,
NULL,
NULL,
sizeof (GstRtpMP4APay),
0,
(GInstanceInitFunc) gst_rtp_mp4a_pay_init,
};
rtpmp4apay_type =
g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4APay",
&rtpmp4apay_info, 0);
}
return rtpmp4apay_type;
}
static void
gst_rtp_mp4a_pay_base_init (GstRtpMP4APayClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4apay_details);
}
static void
gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
gstelement_class->change_state = gst_rtp_mp4a_pay_change_state;
gstbasertppayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpmp4apay_debug, "rtpmp4apay", 0,
"MP4A-LATM RTP Payloader");
}
static void
gst_rtp_mp4a_pay_init (GstRtpMP4APay * rtpmp4apay)
{
rtpmp4apay->rate = 90000;
rtpmp4apay->profile = g_strdup ("1");
}
static void
gst_rtp_mp4a_pay_finalize (GObject * object)
{
GstRtpMP4APay *rtpmp4apay;
rtpmp4apay = GST_RTP_MP4A_PAY (object);
g_free (rtpmp4apay->params);
rtpmp4apay->params = NULL;
if (rtpmp4apay->config)
gst_buffer_unref (rtpmp4apay->config);
rtpmp4apay->config = NULL;
g_free (rtpmp4apay->profile);
rtpmp4apay->profile = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static unsigned sampling_table[16] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
static gboolean
gst_rtp_mp4a_pay_parse_audio_config (GstRtpMP4APay * rtpmp4apay,
GstBuffer * buffer)
{
guint8 *data;
guint size;
guint8 objectType;
guint8 samplingIdx;
guint8 channelCfg;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 2)
goto too_short;
/* any object type is fine, we need to copy it to the profile-level-id field. */
objectType = (data[0] & 0xf8) >> 3;
if (objectType == 0)
goto invalid_object;
samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
/* only fixed values for now */
if (samplingIdx > 12 && samplingIdx != 15)
goto wrong_freq;
channelCfg = ((data[1] & 0x78) >> 3);
if (channelCfg > 7)
goto wrong_channels;
/* rtp rate depends on sampling rate of the audio */
if (samplingIdx == 15) {
if (size < 5)
goto too_short;
/* index of 15 means we get the rate in the next 24 bits */
rtpmp4apay->rate = ((data[1] & 0x7f) << 17) |
((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
} else {
/* else use the rate from the table */
rtpmp4apay->rate = sampling_table[samplingIdx];
}
/* extra rtp params contain the number of channels */
g_free (rtpmp4apay->params);
rtpmp4apay->params = g_strdup_printf ("%d", channelCfg);
/* audio stream type */
rtpmp4apay->streamtype = "5";
/* profile */
g_free (rtpmp4apay->profile);
rtpmp4apay->profile = g_strdup_printf ("%d", objectType);
GST_DEBUG_OBJECT (rtpmp4apay,
"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
samplingIdx, rtpmp4apay->rate, channelCfg);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
(NULL), ("config string too short, expected 2 bytes, got %d", size));
return FALSE;
}
invalid_object:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
(NULL), ("invalid object type 0"));
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d, must < 8", channelCfg));
return FALSE;
}
}
static void
gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
{
gchar *config;
GValue v = { 0 };
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4apay->config);
config = gst_value_serialize (&v);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4apay),
"cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (config);
}
static gboolean
gst_rtp_mp4a_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4APay *rtpmp4apay;
GstStructure *structure;
const GValue *codec_data;
rtpmp4apay = GST_RTP_MP4A_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
codec_data = gst_structure_get_value (structure, "codec_data");
if (codec_data) {
GST_LOG_OBJECT (rtpmp4apay, "got codec_data");
if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
GstBuffer *buffer, *cbuffer;
guint8 *config;
guint8 *data;
guint size, i;
gboolean res;
buffer = gst_value_get_buffer (codec_data);
GST_LOG_OBJECT (rtpmp4apay, "configuring codec_data");
/* parse buffer */
res = gst_rtp_mp4a_pay_parse_audio_config (rtpmp4apay, buffer);
if (!res)
goto config_failed;
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
/* make the StreamMuxConfig, we need 15 bits for the header */
config = g_malloc0 (size + 2);
/* Create StreamMuxConfig according to ISO/IEC 14496-3:
*
* audioMuxVersion == 0 (1 bit)
* allStreamsSameTimeFraming == 1 (1 bit)
* numSubFrames == numSubFrames (6 bits)
* numProgram == 0 (4 bits)
* numLayer == 0 (3 bits)
*/
config[0] = 0x40;
config[1] = 0x00;
/* append the config bits, shifting them 1 bit left */
for (i = 0; i < size; i++) {
config[i + 1] |= ((data[i] & 0x80) >> 7);
config[i + 2] |= ((data[i] & 0x7f) << 1);
}
cbuffer = gst_buffer_new ();
GST_BUFFER_DATA (cbuffer) = config;
GST_BUFFER_MALLOCDATA (cbuffer) = config;
GST_BUFFER_SIZE (cbuffer) = size + 2;
/* now we can configure the buffer */
if (rtpmp4apay->config)
gst_buffer_unref (rtpmp4apay->config);
rtpmp4apay->config = cbuffer;
}
}
gst_basertppayload_set_options (payload, "audio", TRUE, "MP4A-LATM",
rtpmp4apay->rate);
gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
return TRUE;
/* ERRORS */
config_failed:
{
GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
return FALSE;
}
}
/* we expect buffers as exactly one complete AU
*/
static GstFlowReturn
gst_rtp_mp4a_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4APay *rtpmp4apay;
GstFlowReturn ret;
GstBuffer *outbuf;
guint count, mtu, size;
guint8 *data;
gboolean fragmented;
ret = GST_FLOW_OK;
rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
size = GST_BUFFER_SIZE (buffer);
data = GST_BUFFER_DATA (buffer);
fragmented = FALSE;
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4apay);
while (size > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
if (!fragmented) {
/* first packet calculate space for the packet including the header */
count = size;
while (count >= 0xff) {
packet_len++;
count -= 0xff;
}
packet_len++;
}
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, mtu);
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
GST_DEBUG_OBJECT (rtpmp4apay,
"avail %d, towrite %d, packet_len %d, payload_len %d", size, towrite,
packet_len, payload_len);
/* create buffer to hold the payload. */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
if (!fragmented) {
/* first packet write the header */
count = size;
while (count >= 0xff) {
*payload++ = 0xff;
payload_len--;
count -= 0xff;
}
*payload++ = count;
payload_len--;
}
/* copy data to payload */
memcpy (payload, data, payload_len);
data += payload_len;
size -= payload_len;
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (outbuf, size == 0);
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4apay), outbuf);
fragmented = TRUE;
}
return ret;
}
static GstStateChangeReturn
gst_rtp_mp4a_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpMP4APay *rtpmp4apay;
GstStateChangeReturn ret;
rtpmp4apay = GST_RTP_MP4A_PAY (element);
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4apay",
GST_RANK_NONE, GST_TYPE_RTP_MP4A_PAY);
}

63
gst/rtp/gstrtpmp4apay.h Normal file
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@ -0,0 +1,63 @@
/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_RTP_MP4A_PAY_H__
#define __GST_RTP_MP4A_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_MP4A_PAY \
(gst_rtp_mp4a_pay_get_type())
#define GST_RTP_MP4A_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_MP4A_PAY,GstRtpMP4APay))
#define GST_RTP_MP4A_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_MP4A_PAY,GstRtpMP4APayClass))
#define GST_IS_RTP_MP4A_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_MP4A_PAY))
#define GST_IS_RTP_MP4A_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_MP4A_PAY))
typedef struct _GstRtpMP4APay GstRtpMP4APay;
typedef struct _GstRtpMP4APayClass GstRtpMP4APayClass;
struct _GstRtpMP4APay
{
GstBaseRTPPayload payload;
gint rate;
gchar *params;
gchar *profile;
const gchar *streamtype;
GstBuffer *config;
};
struct _GstRtpMP4APayClass
{
GstBaseRTPPayloadClass parent_class;
};
gboolean gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_MP4A_PAY_H__ */