gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4apay.c
Sebastian Dröge b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00

463 lines
13 KiB
C

/* 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 "gstrtpelements.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);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4apay, "rtpmp4apay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4A_PAY, rtp_element_init (plugin));
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_static_pad_template (gstelement_class,
&gst_rtp_mp4a_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&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_base_payload_allocate_output_buffer (basepayload,
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);
if (size == payload_len)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
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_audio_meta (rtpmp4apay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
gst_buffer_list_add (list, outbuf);
offset += payload_len;
size -= payload_len;
/* copy incoming 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;
}