gstreamer/gst/rtp/gstrtpmp4adepay.c
Stefan Kost 988f228da7 rtpmp4adepay: grab the sampling arte and put into caps
This is needed to be able to mux the received audio into mp4 (in the case of
aac). Fixes #625825.
2010-09-06 21:54:25 +03:00

376 lines
11 KiB
C

/* GStreamer
* Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
* <2007> 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 version 2 as published by the Free Software Foundation.
*
* 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 <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstrtpmp4adepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg,"
"mpegversion = (int) 4," "framed = (boolean) true, "
"stream-format = (string) raw")
);
static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
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
*
* "profile-level-id=[1,MAX]"
* "config="
*/
)
);
GST_BOILERPLATE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void gst_rtp_mp4a_depay_finalize (GObject * object);
static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_mp4a_depay_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_mp4a_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
gst_element_class_set_details_simple (element_class,
"RTP MPEG4 audio depayloader", "Codec/Depayloader/Network",
"Extracts MPEG4 audio from RTP packets (RFC 3016)",
"Nokia Corporation (contact <stefan.kost@nokia.com>), "
"Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
gstbasertpdepayload_class->process = gst_rtp_mp4a_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
"MPEG4 audio RTP Depayloader");
}
static void
gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay,
GstRtpMP4ADepayClass * klass)
{
rtpmp4adepay->adapter = gst_adapter_new ();
}
static void
gst_rtp_mp4a_depay_finalize (GObject * object)
{
GstRtpMP4ADepay *rtpmp4adepay;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
g_object_unref (rtpmp4adepay->adapter);
rtpmp4adepay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4ADepay *rtpmp4adepay;
GstCaps *srccaps;
const gchar *str;
gint clock_rate;
gint object_type;
gint channels = 2; /* default */
gboolean res;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
if (!gst_structure_get_int (structure, "object", &object_type))
object_type = 2; /* AAC LC default */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4,
"framed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, channels,
"stream-format", G_TYPE_STRING, "raw", NULL);
if ((str = gst_structure_get_string (structure, "config"))) {
GValue v = { 0 };
g_value_init (&v, GST_TYPE_BUFFER);
if (gst_value_deserialize (&v, str)) {
GstBuffer *buffer;
guint8 *data;
guint size;
gint i;
guint sr_idx;
static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000
};
buffer = gst_value_get_buffer (&v);
gst_buffer_ref (buffer);
g_value_unset (&v);
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 2) {
GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", size);
goto bad_config;
}
/* Parse StreamMuxConfig according to ISO/IEC 14496-3:
*
* audioMuxVersion == 0 (1 bit)
* allStreamsSameTimeFraming == 1 (1 bit)
* numSubFrames == rtpmp4adepay->numSubFrames (6 bits)
* numProgram == 0 (4 bits)
* numLayer == 0 (3 bits)
*
* We only require audioMuxVersion == 0;
*
* The remaining bit of the second byte and the rest of the bits are used
* for audioSpecificConfig which we need to set in codec_info.
*/
if ((data[0] & 0x80) != 0x00) {
GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
goto bad_config;
}
rtpmp4adepay->numSubFrames = (data[0] & 0x3F);
GST_LOG_OBJECT (rtpmp4adepay, "numSubFrames %d",
rtpmp4adepay->numSubFrames);
/* shift rest of string 15 bits down */
size -= 2;
for (i = 0; i < size; i++) {
data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
}
/* grab and set sampling rate */
sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
if (sr_idx < G_N_ELEMENTS (aac_sample_rates)) {
gst_caps_set_simple (srccaps,
"rate", G_TYPE_INT, (gint) aac_sample_rates[sr_idx], NULL);
} else {
GST_WARNING ("Invalid sample rate index %u", sr_idx);
}
/* ignore remaining bit, we're only interested in full bytes */
GST_BUFFER_SIZE (buffer) = size;
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, buffer, NULL);
gst_buffer_unref (buffer);
} else {
g_warning ("cannot convert config to buffer");
}
}
bad_config:
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
}
static GstBuffer *
gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4ADepay *rtpmp4adepay;
GstBuffer *outbuf;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
gst_adapter_clear (rtpmp4adepay->adapter);
}
outbuf = gst_rtp_buffer_get_payload_buffer (buf);
gst_adapter_push (rtpmp4adepay->adapter, outbuf);
/* RTP marker bit indicates the last packet of the AudioMuxElement => create
* and push a buffer */
if (gst_rtp_buffer_get_marker (buf)) {
guint avail;
guint i;
guint8 *data;
guint pos;
avail = gst_adapter_available (rtpmp4adepay->adapter);
GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail);
outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
data = GST_BUFFER_DATA (outbuf);
/* position in data we are at */
pos = 0;
/* looping through the number of sub-frames in the audio payload */
for (i = 0; i <= rtpmp4adepay->numSubFrames; i++) {
/* determine payload length and set buffer data pointer accordingly */
guint skip;
guint data_len;
guint32 timestamp;
GstBuffer *tmp = NULL;
timestamp = gst_rtp_buffer_get_timestamp (buf);
/* each subframe starts with a variable length encoding */
data_len = 0;
for (skip = 0; skip < avail; skip++) {
data_len += data[skip];
if (data[skip] != 0xff)
break;
}
skip++;
/* this can not be possible, we have not enough data or the length
* decoding failed because we ran out of data. */
if (skip + data_len > avail)
goto wrong_size;
GST_LOG_OBJECT (rtpmp4adepay,
"subframe %u, header len %u, data len %u, left %u", i, skip, data_len,
avail);
/* take data out, skip the header */
pos += skip;
tmp = gst_buffer_create_sub (outbuf, pos, data_len);
/* skip data too */
skip += data_len;
pos += data_len;
/* update our pointers whith what we consumed */
data += skip;
avail -= skip;
gst_buffer_set_caps (tmp, GST_PAD_CAPS (depayload->srcpad));
/* only apply the timestamp for the first buffer. Based on gstrtpmp4gdepay.c */
if (i == 0)
gst_base_rtp_depayload_push_ts (depayload, timestamp, tmp);
else
gst_base_rtp_depayload_push (depayload, tmp);
}
/* just a check that lengths match */
if (avail) {
GST_ELEMENT_WARNING (depayload, STREAM, DECODE,
("Packet invalid"), ("Not all payload consumed: "
"possible wrongly encoded packet."));
}
gst_buffer_unref (outbuf);
}
return NULL;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
("Packet did not validate"), ("wrong packet size"));
gst_buffer_unref (outbuf);
return NULL;
}
}
static GstStateChangeReturn
gst_rtp_mp4a_depay_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpMP4ADepay *rtpmp4adepay;
GstStateChangeReturn ret;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_adapter_clear (rtpmp4adepay->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4adepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_MP4A_DEPAY);
}