blob: b42cffdc8f87d8b06c1ae9173b06ef8f0acea8f5 [file] [log] [blame]
/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpmp4apay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4apay_debug);
#define GST_CAT_DEFAULT (rtpmp4apay_debug)
/* FIXME: add framed=(boolean)true once our encoders have this field set
* on their output caps */
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, "
"stream-format=(string)raw")
);
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_finalize (GObject * object);
static gboolean gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
#define gst_rtp_mp4a_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMP4APay, gst_rtp_mp4a_pay, GST_TYPE_RTP_BASE_PAYLOAD)
static void gst_rtp_mp4a_pay_class_init (GstRtpMP4APayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4a_pay_finalize;
gstrtpbasepayload_class->set_caps = gst_rtp_mp4a_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4a_pay_handle_buffer;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_mp4a_pay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_mp4a_pay_sink_template));
gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG4 audio payloader", "Codec/Payloader/Network/RTP",
"Payload MPEG4 audio as RTP packets (RFC 3016)",
"Wim Taymans <wim.taymans@gmail.com>");
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 const unsigned int 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)
{
GstMapInfo map;
guint8 *data;
gsize size;
guint8 objectType;
guint8 samplingIdx;
guint8 channelCfg;
gst_buffer_map (buffer, &map, GST_MAP_READ);
data = map.data;
size = map.size;
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);
gst_buffer_unmap (buffer, &map);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
(NULL),
("config string too short, expected 2 bytes, got %" G_GSIZE_FORMAT,
size));
gst_buffer_unmap (buffer, &map);
return FALSE;
}
invalid_object:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, FORMAT,
(NULL), ("invalid object type 0"));
gst_buffer_unmap (buffer, &map);
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
gst_buffer_unmap (buffer, &map);
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4apay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d, must < 8", channelCfg));
gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
static gboolean
gst_rtp_mp4a_pay_new_caps (GstRtpMP4APay * rtpmp4apay)
{
gchar *config;
GValue v = { 0 };
gboolean res;
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4apay->config);
config = gst_value_serialize (&v);
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4apay),
"cpresent", G_TYPE_STRING, "0", "config", G_TYPE_STRING, config, NULL);
g_value_unset (&v);
g_free (config);
return res;
}
static gboolean
gst_rtp_mp4a_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpMP4APay *rtpmp4apay;
GstStructure *structure;
const GValue *codec_data;
gboolean res, framed = TRUE;
const gchar *stream_format;
rtpmp4apay = GST_RTP_MP4A_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
/* this is already handled by the template caps, but it is better
* to leave here to have meaningful warning messages when linking
* fails */
stream_format = gst_structure_get_string (structure, "stream-format");
if (stream_format) {
if (strcmp (stream_format, "raw") != 0) {
GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format must be 'raw', "
"%s is not supported", stream_format);
return FALSE;
}
} else {
GST_WARNING_OBJECT (rtpmp4apay, "AAC's stream-format not specified, "
"assuming 'raw'");
}
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;
GstMapInfo map;
GstMapInfo cmap;
guint i;
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;
gst_buffer_map (buffer, &map, GST_MAP_READ);
/* make the StreamMuxConfig, we need 15 bits for the header */
cbuffer = gst_buffer_new_and_alloc (map.size + 2);
gst_buffer_map (cbuffer, &cmap, GST_MAP_WRITE);
memset (cmap.data, 0, map.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)
*/
cmap.data[0] = 0x40;
cmap.data[1] = 0x00;
/* append the config bits, shifting them 1 bit left */
for (i = 0; i < map.size; i++) {
cmap.data[i + 1] |= ((map.data[i] & 0x80) >> 7);
cmap.data[i + 2] |= ((map.data[i] & 0x7f) << 1);
}
gst_buffer_unmap (cbuffer, &cmap);
gst_buffer_unmap (buffer, &map);
/* now we can configure the buffer */
if (rtpmp4apay->config)
gst_buffer_unref (rtpmp4apay->config);
rtpmp4apay->config = cbuffer;
}
}
if (gst_structure_get_boolean (structure, "framed", &framed) && !framed) {
GST_WARNING_OBJECT (payload, "Need framed AAC data as input!");
}
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MP4A-LATM",
rtpmp4apay->rate);
res = gst_rtp_mp4a_pay_new_caps (rtpmp4apay);
return res;
/* ERRORS */
config_failed:
{
GST_DEBUG_OBJECT (rtpmp4apay, "failed to parse config");
return FALSE;
}
}
#define RTP_HEADER_LEN 12
/* we expect buffers as exactly one complete AU
*/
static GstFlowReturn
gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4APay *rtpmp4apay;
GstFlowReturn ret;
GstBufferList *list;
guint mtu;
guint offset;
gsize size;
gboolean fragmented;
GstClockTime timestamp;
ret = GST_FLOW_OK;
rtpmp4apay = GST_RTP_MP4A_PAY (basepayload);
offset = 0;
size = gst_buffer_get_size (buffer);
timestamp = GST_BUFFER_PTS (buffer);
fragmented = FALSE;
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4apay);
list = gst_buffer_list_new_sized (size / (mtu - RTP_HEADER_LEN) + 1);
while (size > 0) {
guint towrite;
GstBuffer *outbuf;
guint payload_len;
guint packet_len;
guint header_len;
GstBuffer *paybuf;
GstRTPBuffer rtp = { NULL };
header_len = 0;
if (!fragmented) {
guint count;
/* first packet calculate space for the packet including the header */
count = size;
while (count >= 0xff) {
header_len++;
count -= 0xff;
}
header_len++;
}
packet_len = gst_rtp_buffer_calc_packet_len (header_len + size, 0, 0);
towrite = MIN (packet_len, mtu);
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
payload_len -= header_len;
GST_DEBUG_OBJECT (rtpmp4apay,
"avail %" G_GSIZE_FORMAT
", header_len %d, packet_len %d, payload_len %d", size, header_len,
packet_len, payload_len);
/* create buffer to hold the payload. */
outbuf = gst_rtp_buffer_new_allocate (header_len, 0, 0);
/* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
if (!fragmented) {
guint8 *payload = gst_rtp_buffer_get_payload (&rtp);
guint count;
/* first packet write the header */
count = size;
while (count >= 0xff) {
*payload++ = 0xff;
count -= 0xff;
}
*payload++ = count;
}
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (&rtp, size == payload_len);
gst_rtp_buffer_unmap (&rtp);
/* create a new buf to hold the payload */
paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL,
offset, payload_len);
/* join memory parts */
gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4apay), outbuf, paybuf,
g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
outbuf = gst_buffer_append (outbuf, paybuf);
gst_buffer_list_add (list, outbuf);
offset += payload_len;
size -= payload_len;
/* copy incomming timestamp (if any) to outgoing buffers */
GST_BUFFER_PTS (outbuf) = timestamp;
fragmented = TRUE;
}
ret =
gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4apay), list);
gst_buffer_unref (buffer);
return ret;
}
gboolean
gst_rtp_mp4a_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4apay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY);
}