gstreamer/subprojects/gst-plugins-bad/gst/mpegtsmux/gstatscmux.c

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/* ATSC Transport Stream muxer
* Copyright (C) 2019 Mathieu Duponchelle <mathieu@centricular.com>
*
* atscmux.c:
*
* 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.
*
* SPDX-License-Identifier: LGPL-2.0-or-later
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*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
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#include "gstatscmux.h"
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GST_DEBUG_CATEGORY (gst_atsc_mux_debug);
#define GST_CAT_DEFAULT gst_atsc_mux_debug
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G_DEFINE_TYPE (GstATSCMux, gst_atsc_mux, GST_TYPE_BASE_TS_MUX);
GST_ELEMENT_REGISTER_DEFINE (atscmux, "atscmux", GST_RANK_PRIMARY,
gst_atsc_mux_get_type ());
#define parent_class gst_atsc_mux_parent_class
#define ATSCMUX_ST_PS_AUDIO_EAC3 0x87
static GstStaticPadTemplate gst_atsc_mux_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpegts, "
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"systemstream = (boolean) true, " "packetsize = (int) 188 ")
);
static GstStaticPadTemplate gst_atsc_mux_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("video/mpeg, "
"parsed = (boolean) TRUE, "
"mpegversion = (int) 2, "
"systemstream = (boolean) false; "
"video/x-h264,stream-format=(string)byte-stream,"
"alignment=(string){au, nal}; "
"audio/x-ac3, framed = (boolean) TRUE;"
"audio/x-eac3, framed = (boolean) TRUE;"));
/* Internals */
static void
gst_atsc_mux_stream_get_es_descrs (TsMuxStream * stream,
GstMpegtsPMTStream * pmt_stream, gpointer user_data)
{
GstMpegtsDescriptor *descriptor;
if (stream->stream_type == ATSCMUX_ST_PS_AUDIO_EAC3) {
guint8 add_info[4];
guint8 *pos;
pos = add_info;
/* audio_stream_descriptor () | ATSC A/52-2018 Annex G
*
* descriptor_tag 8 uimsbf
* descriptor_length 8 uimsbf
* reserved 1 '1'
* bsid_flag 1 bslbf
* mainid_flag 1 bslbf
* asvc_flag 1 bslbf
* mixinfoexists 1 bslbf
* substream1_flag 1 bslbf
* substream2_flag 1 bslbf
* substream3_flag 1 bslbf
* reserved 1 '1'
* full_service_flag 1 bslbf
* audio_service_type 3 uimsbf
* number_of_channels 3 uimsbf
* [...]
*/
*pos++ = 0xCC;
*pos++ = 2;
/* 1 bit reserved, all other flags unset */
*pos++ = 0x80;
/* 1 bit reserved,
* 1 bit set for full_service_flag,
* 3 bits hardcoded audio_service_type "Complete Main",
* 3 bits number_of_channels
*/
switch (stream->audio_channels) {
case 1:
*pos++ = 0xC0; /* Mono */
break;
case 2:
*pos++ = 0xC0 | 0x2; /* 2-channel (stereo) */
break;
case 3:
case 4:
case 5:
*pos++ = 0xC0 | 0x4; /* Multichannel audio (> 2 channels; <= 3/2 + LFE channels) */
break;
case 6:
default:
*pos++ = 0xC0 | 0x5; /* Multichannel audio(> 3/2 + LFE channels) */
}
descriptor = gst_mpegts_descriptor_from_registration ("EAC3", add_info, 4);
g_ptr_array_add (pmt_stream->descriptors, descriptor);
descriptor =
gst_mpegts_descriptor_from_custom (GST_MTS_DESC_ATSC_EAC3, add_info, 4);
g_ptr_array_add (pmt_stream->descriptors, descriptor);
} else if (stream->stream_type == TSMUX_ST_PS_AUDIO_AC3) {
int wr_size = 0;
guint8 *add_info = NULL;
guint8 data;
guint bitrate;
gboolean has_language;
guint bitrates[20][2] = {
{32000, 0x00}
,
{40000, 0x01}
,
{48000, 0x02}
,
{56000, 0x03}
,
{64000, 0x04}
,
{80000, 0x05}
,
{96000, 0x06}
,
{112000, 0x07}
,
{128000, 0x08}
,
{160000, 0x09}
,
{192000, 0x0A}
,
{224000, 0x0B}
,
{256000, 0x0C}
,
{320000, 0x0D}
,
{384000, 0x0E}
,
{448000, 0x0F}
,
{512000, 0x10}
,
{576000, 0x11}
,
{640000, 0x12}
};
gint i;
guint bitrate_code = 0x12;
GstByteWriter writer;
gst_byte_writer_init_with_size (&writer, 7, FALSE);
/* audio_stream_descriptor () | ATSC A/52-2001 Annex A
*
* descriptor_tag 8 uimsbf
* descriptor_length 8 uimsbf
* sample_rate_code 3 bslbf
* bsid 5 bslbf
* bit_rate_code 6 bslbf
* surround_mode 2 bslbf
* bsmod 3 bslbf
* num_channels 4 bslbf
* full_svc 1 bslbf
* langcod 8 bslbf
* mainid 3 uimsbf
* priority 2 bslbf
* reserved 3 '111'
* textlen 7 uimsbf
* text_code 1 bslbf
* text 8*textlen bslbf
* language_flag 1 bslbf
* language_flag_2 1 bslbf
* reserved 6 '111111'
* language if flag 3*8 uimbsf
* language_2 if flag_2 3*8 uimsbf
*/
/* 3 bits sample_rate_code, 5 bits hardcoded bsid (default ver 8) */
switch (stream->audio_sampling) {
case 48000:
data = 0x08;
break;
case 44100:
data = 0x28;
break;
case 32000:
data = 0x48;
break;
default:
data = 0xE8;
break; /* 48, 44.1 or 32 Khz */
}
gst_byte_writer_put_uint8 (&writer, data);
/* 1 bit bit_rate_limit, 5 bits bit_rate_code, 2 bits suround_mode */
bitrate = MAX (stream->audio_bitrate, stream->max_bitrate);
for (i = 0; i < G_N_ELEMENTS (bitrates); i++) {
if (bitrate < bitrates[i][0]) {
break;
}
bitrate_code = bitrates[i][1];
}
data = bitrate_code << 2;
data |= 0x80; /* This is a maximum bitrate */
gst_byte_writer_put_uint8 (&writer, data);
/* 3 bits bsmod, 4 bits num_channels, 1 bit full_svc */
switch (stream->audio_channels) {
case 1:
data = 0x01 << 1;
break; /* 1/0 */
case 2:
data = 0x02 << 1;
break; /* 2/0 */
case 3:
data = 0x0A << 1;
break; /* <= 3 */
case 4:
data = 0x0B << 1;
break; /* <= 4 */
case 5:
data = 0x0C << 1;
break; /* <= 5 */
case 6:
default:
data = 0x0D << 1;
break; /* <= 6 */
}
data |= 0x01; /* full_svc is hardcoded to 1 for now */
gst_byte_writer_put_uint8 (&writer, data);
/* deprecated langcod */
data = 0xff;
gst_byte_writer_put_uint8 (&writer, data);
/* langcod2 skipped because num_channels > 0 (no dual mono) */
/* 3 bits mainid, 2 bits priority, 3 bits reserved */
data = 0x0f;
gst_byte_writer_put_uint8 (&writer, data);
/* 7 bits textlen, 1 bit text_code */
data = 0x00;
gst_byte_writer_put_uint8 (&writer, data);
/* no text provided, jumping directly to language */
has_language = (stream->language[0] != '\0');
if (has_language) {
data = 0xbf;
gst_byte_writer_put_uint8 (&writer, data);
gst_byte_writer_put_data (&writer, (guint8 *) stream->language, 3);
} else {
data = 0x3f;
gst_byte_writer_put_uint8 (&writer, data);
}
descriptor = gst_mpegts_descriptor_from_registration ("AC-3", NULL, 0);
g_ptr_array_add (pmt_stream->descriptors, descriptor);
wr_size = gst_byte_writer_get_size (&writer);
add_info = gst_byte_writer_reset_and_get_data (&writer);
descriptor =
gst_mpegts_descriptor_from_custom (GST_MTS_DESC_AC3_AUDIO_STREAM,
add_info, wr_size);
g_ptr_array_add (pmt_stream->descriptors, descriptor);
} else {
tsmux_stream_default_get_es_descrs (stream, pmt_stream);
}
}
static TsMuxStream *
gst_atsc_mux_create_new_stream (guint16 new_pid, TsMuxStreamType stream_type,
guint stream_number, gpointer user_data)
{
TsMuxStream *ret = tsmux_stream_new (new_pid, stream_type, stream_number);
if (stream_type == ATSCMUX_ST_PS_AUDIO_EAC3) {
ret->id = 0xBD;
ret->pi.flags |= TSMUX_PACKET_FLAG_PES_FULL_HEADER;
ret->gst_stream_type = GST_STREAM_TYPE_AUDIO;
} else if (stream_type == TSMUX_ST_PS_AUDIO_AC3) {
ret->id = 0xBD;
ret->id_extended = 0;
ret->gst_stream_type = GST_STREAM_TYPE_AUDIO;
}
tsmux_stream_set_get_es_descriptors_func (ret,
gst_atsc_mux_stream_get_es_descrs, user_data);
return ret;
}
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/* GstBaseTsMux implementation */
static TsMux *
gst_atsc_mux_create_ts_mux (GstBaseTsMux * mpegtsmux)
{
TsMux *ret = ((GstBaseTsMuxClass *) parent_class)->create_ts_mux (mpegtsmux);
GstMpegtsAtscMGT *mgt;
GstMpegtsAtscSTT *stt;
GstMpegtsAtscRRT *rrt;
GstMpegtsSection *section;
mgt = gst_mpegts_atsc_mgt_new ();
section = gst_mpegts_section_from_atsc_mgt (mgt);
tsmux_add_mpegts_si_section (ret, section);
stt = gst_mpegts_atsc_stt_new ();
section = gst_mpegts_section_from_atsc_stt (stt);
tsmux_add_mpegts_si_section (ret, section);
rrt = gst_mpegts_atsc_rrt_new ();
section = gst_mpegts_section_from_atsc_rrt (rrt);
tsmux_add_mpegts_si_section (ret, section);
tsmux_set_new_stream_func (ret, gst_atsc_mux_create_new_stream, mpegtsmux);
return ret;
}
static guint
gst_atsc_mux_handle_media_type (GstBaseTsMux * mux, const gchar * media_type,
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GstBaseTsMuxPad * pad)
{
guint ret = TSMUX_ST_RESERVED;
if (!g_strcmp0 (media_type, "audio/x-eac3")) {
ret = ATSCMUX_ST_PS_AUDIO_EAC3;
}
return ret;
}
static void
gst_atsc_mux_class_init (GstATSCMuxClass * klass)
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{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseTsMuxClass *mpegtsmux_class = (GstBaseTsMuxClass *) klass;
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GST_DEBUG_CATEGORY_INIT (gst_atsc_mux_debug, "atscmux", 0, "ATSC muxer");
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gst_element_class_set_static_metadata (gstelement_class,
"ATSC Transport Stream Muxer", "Codec/Muxer",
"Multiplexes media streams into an ATSC-compliant Transport Stream",
"Mathieu Duponchelle <mathieu@centricular.com>");
mpegtsmux_class->create_ts_mux = gst_atsc_mux_create_ts_mux;
mpegtsmux_class->handle_media_type = gst_atsc_mux_handle_media_type;
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&gst_atsc_mux_sink_factory, GST_TYPE_BASE_TS_MUX_PAD);
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gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&gst_atsc_mux_src_factory, GST_TYPE_AGGREGATOR_PAD);
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}
static void
gst_atsc_mux_init (GstATSCMux * mux)
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{
}