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rtp: ldacpay: Add LDAC RTP payloader

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
This commit is contained in:
Sanchayan Maity 2020-09-14 13:12:50 +05:30 committed by Sanchayan Maity
parent e3c16d0194
commit 8c3ec64473
5 changed files with 262 additions and 0 deletions
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27
docs/gst_plugins_cache.json
View file

@ -14315,6 +14315,33 @@
"properties": {},
"rank": "secondary"
},
"rtpldacpay": {
"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
"description": "Payload LDAC audio as RTP packets",
"hierarchy": [
"GstRtpLdacPay",
"GstRTPBasePayload",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network",
"long-name": "RTP packet payloader",
"pad-templates": {
"sink": {
"caps": "audio/x-ldac:\n channels: [ 1, 2 ]\n rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: audio\n payload: [ 96, 127 ]\n clock-rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n encoding-name: X-GST-LDAC\n",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
"rtpmp1sdepay": {
"author": "Wim Taymans <wim.taymans@gmail.com>",
"description": "Extracts MPEG1 System Streams from RTP packets (RFC 3555)",

4
gst/rtp/gstrtp.c
View file

@ -84,6 +84,7 @@
#include "gstrtpL16pay.h"
#include "gstrtpL24depay.h"
#include "gstrtpL24pay.h"
#include "gstrtpldacpay.h"
#include "gstasteriskh263.h"
#include "gstrtpmp1sdepay.h"
#include "gstrtpmp2tdepay.h"
@ -302,6 +303,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_L24_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_ldac_pay_plugin_init (plugin))
return FALSE;
if (!gst_asteriskh263_plugin_init (plugin))
return FALSE;

175
gst/rtp/gstrtpldacpay.c Normal file
View file

@ -0,0 +1,175 @@
/* GStreamer RTP LDAC payloader
* Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
*
* 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.
*/
/**
* SECTION:element-rtpldacpay
* @title: rtpldacpay
*
* Payload LDAC encoded audio into RTP packets.
*
* LDAC does not have a public specification and concerns itself only with
* bluetooth transmission. Due to the unavailability of a specification, we
* consider the encoding-name as X-GST-LDAC.
*
* The best reference is [libldac](https://android.googlesource.com/platform/external/libldac/)
* and the A2DP LDAC implementation in Android's bluetooth stack [Flouride]
* (https://android.googlesource.com/platform/system/bt/+/refs/heads/master/stack/a2dp/a2dp_vendor_ldac_encoder.cc).
*
* ## Example pipeline
* |[
* gst-launch-1.0 -v audiotestsrc ! ldacenc ! rtpldacpay mtu=679 ! avdtpsink
* ]| This example pipeline will payload LDAC encoded audio.
*
* Since: 1.20
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/audio/audio.h>
#include "gstrtpldacpay.h"
#include "gstrtputils.h"
#define GST_RTP_HEADER_LENGTH 12
/* MTU size required for LDAC A2DP streaming */
#define GST_LDAC_MTU_REQUIRED 679
GST_DEBUG_CATEGORY_STATIC (gst_rtp_ldac_pay_debug);
#define GST_CAT_DEFAULT gst_rtp_ldac_pay_debug
#define parent_class gst_rtp_ldac_pay_parent_class
G_DEFINE_TYPE (GstRtpLdacPay, gst_rtp_ldac_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static GstStaticPadTemplate gst_rtp_ldac_pay_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-ldac, "
"channels = (int) [ 1, 2 ], "
"rate = (int) { 44100, 48000, 88200, 96000 }")
);
static GstStaticPadTemplate gst_rtp_ldac_pay_src_factory =
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) { 44100, 48000, 88200, 96000 },"
"encoding-name = (string) \"X-GST-LDAC\"")
);
static gboolean gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
static void
gst_rtp_ldac_pay_class_init (GstRtpLdacPayClass * klass)
{
GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
payload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_ldac_pay_set_caps);
payload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_rtp_ldac_pay_handle_buffer);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_ldac_pay_sink_factory);
gst_element_class_add_static_pad_template (element_class,
&gst_rtp_ldac_pay_src_factory);
gst_element_class_set_static_metadata (element_class, "RTP packet payloader",
"Codec/Payloader/Network", "Payload LDAC audio as RTP packets",
"Sanchayan Maity <sanchayan@asymptotic.io>");
GST_DEBUG_CATEGORY_INIT (gst_rtp_ldac_pay_debug, "rtpldacpay", 0,
"RTP LDAC payloader");
}
static void
gst_rtp_ldac_pay_init (GstRtpLdacPay * self)
{
}
static gboolean
gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
GstStructure *structure;
gint rate;
if (GST_RTP_BASE_PAYLOAD_MTU (ldacpay) < GST_LDAC_MTU_REQUIRED) {
GST_ERROR_OBJECT (ldacpay, "Invalid MTU %d, should be >= %d",
GST_RTP_BASE_PAYLOAD_MTU (ldacpay), GST_LDAC_MTU_REQUIRED);
return FALSE;
}
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate)) {
GST_ERROR_OBJECT (ldacpay, "Failed to get audio rate from caps");
return FALSE;
}
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "X-GST-LDAC", rate);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}
/*
* LDAC encoder does not handle split frames. Currently, the encoder will
* always emit 660 bytes worth of payload encapsulating multiple LDAC frames.
* This is as per eqmid and GST_LDAC_MTU_REQUIRED passed for configuring the
* encoder upstream. Since the encoder always emit full frames and we do not
* need to handle frame splitting, we do not use an adapter and also push out
* the buffer as it is received.
*/
static GstFlowReturn
gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
GstBuffer *outbuf;
GstClockTime outbuf_frame_duration, outbuf_pts;
gsize buf_sz;
outbuf =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(ldacpay), GST_RTP_HEADER_LENGTH, 0, 0);
outbuf_pts = GST_BUFFER_PTS (buffer);
outbuf_frame_duration = GST_BUFFER_DURATION (buffer);
buf_sz = gst_buffer_get_size (buffer);
gst_rtp_copy_audio_meta (ldacpay, outbuf, buffer);
outbuf = gst_buffer_append (outbuf, buffer);
GST_BUFFER_PTS (outbuf) = outbuf_pts;
GST_BUFFER_DURATION (outbuf) = outbuf_frame_duration;
GST_DEBUG_OBJECT (ldacpay,
"Pushing %" G_GSIZE_FORMAT " bytes: %" GST_TIME_FORMAT, buf_sz,
GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (ldacpay), outbuf);
}
gboolean
gst_rtp_ldac_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpldacpay", GST_RANK_NONE,
GST_TYPE_RTP_LDAC_PAY);
}

55
gst/rtp/gstrtpldacpay.h Normal file
View file

@ -0,0 +1,55 @@
/* GStreamer RTP LDAC payloader
* Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
*
* 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.
*/
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/rtp/gstrtpbuffer.h>
G_BEGIN_DECLS
#define GST_TYPE_RTP_LDAC_PAY \
(gst_rtp_ldac_pay_get_type())
#define GST_RTP_LDAC_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_LDAC_PAY,\
GstRtpLdacPay))
#define GST_RTP_LDAC_PAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_LDAC_PAY,\
GstRtpLdacPayClass))
#define GST_IS_RTP_LDAC_PAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_LDAC_PAY))
#define GST_IS_RTP_LDAC_PAY_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_LDAC_PAY))
typedef struct _GstRtpLdacPay GstRtpLdacPay;
typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass;
struct _GstRtpLdacPay {
GstRTPBasePayload base;
};
struct _GstRtpLdacPayClass {
GstRTPBasePayloadClass parent_class;
};
GType gst_rtp_ldac_pay_get_type(void);
gboolean gst_rtp_ldac_pay_plugin_init (GstPlugin * plugin);
G_END_DECLS

1
gst/rtp/meson.build
View file

@ -61,6 +61,7 @@ rtp_sources = [
'gstrtpL16pay.c',
'gstrtpL24depay.c',
'gstrtpL24pay.c',
'gstrtpldacpay.c',
'gstasteriskh263.c',
'gstrtpmp1sdepay.c',
'gstrtpmp2tdepay.c',