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rtp: ldacpay: Add LDAC RTP payloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
This commit is contained in:
Sanchayan Maity
Sanchayan Maity
parent
e3c16d0194
commit
8c3ec64473
5 changed files with 262 additions and 0 deletions
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@ -14315,6 +14315,33 @@
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"properties": {},
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"properties": {},
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"rank": "secondary"
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"rank": "secondary"
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},
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},
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"rtpldacpay": {
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"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
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"description": "Payload LDAC audio as RTP packets",
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"hierarchy": [
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"GstRtpLdacPay",
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"GstRTPBasePayload",
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"GstElement",
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"GstObject",
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"GInitiallyUnowned",
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"GObject"
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],
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"klass": "Codec/Payloader/Network",
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"long-name": "RTP packet payloader",
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"pad-templates": {
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"sink": {
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"caps": "audio/x-ldac:\n channels: [ 1, 2 ]\n rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n",
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"direction": "sink",
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"presence": "always"
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},
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"src": {
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"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",
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"direction": "src",
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"presence": "always"
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}
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},
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"rank": "none"
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},
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"rtpmp1sdepay": {
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"rtpmp1sdepay": {
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"author": "Wim Taymans <wim.taymans@gmail.com>",
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"author": "Wim Taymans <wim.taymans@gmail.com>",
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"description": "Extracts MPEG1 System Streams from RTP packets (RFC 3555)",
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"description": "Extracts MPEG1 System Streams from RTP packets (RFC 3555)",
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@ -84,6 +84,7 @@
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#include "gstrtpL16pay.h"
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#include "gstrtpL16pay.h"
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#include "gstrtpL24depay.h"
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#include "gstrtpL24depay.h"
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#include "gstrtpL24pay.h"
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#include "gstrtpL24pay.h"
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#include "gstrtpldacpay.h"
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#include "gstasteriskh263.h"
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#include "gstasteriskh263.h"
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#include "gstrtpmp1sdepay.h"
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#include "gstrtpmp1sdepay.h"
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#include "gstrtpmp2tdepay.h"
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#include "gstrtpmp2tdepay.h"
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@ -302,6 +303,9 @@ plugin_init (GstPlugin * plugin)
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if (!gst_rtp_L24_depay_plugin_init (plugin))
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if (!gst_rtp_L24_depay_plugin_init (plugin))
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return FALSE;
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return FALSE;
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if (!gst_rtp_ldac_pay_plugin_init (plugin))
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return FALSE;
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if (!gst_asteriskh263_plugin_init (plugin))
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if (!gst_asteriskh263_plugin_init (plugin))
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return FALSE;
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return FALSE;
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175
gst/rtp/gstrtpldacpay.c
Normal file
175
gst/rtp/gstrtpldacpay.c
Normal file
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@ -0,0 +1,175 @@
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/* GStreamer RTP LDAC payloader
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* Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpldacpay
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* @title: rtpldacpay
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*
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* Payload LDAC encoded audio into RTP packets.
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*
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* LDAC does not have a public specification and concerns itself only with
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* bluetooth transmission. Due to the unavailability of a specification, we
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* consider the encoding-name as X-GST-LDAC.
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*
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* The best reference is [libldac](https://android.googlesource.com/platform/external/libldac/)
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* and the A2DP LDAC implementation in Android's bluetooth stack [Flouride]
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* (https://android.googlesource.com/platform/system/bt/+/refs/heads/master/stack/a2dp/a2dp_vendor_ldac_encoder.cc).
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*
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* ## Example pipeline
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* |[
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* gst-launch-1.0 -v audiotestsrc ! ldacenc ! rtpldacpay mtu=679 ! avdtpsink
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* ]| This example pipeline will payload LDAC encoded audio.
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*
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* Since: 1.20
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <gst/audio/audio.h>
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#include "gstrtpldacpay.h"
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#include "gstrtputils.h"
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#define GST_RTP_HEADER_LENGTH 12
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/* MTU size required for LDAC A2DP streaming */
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#define GST_LDAC_MTU_REQUIRED 679
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_ldac_pay_debug);
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#define GST_CAT_DEFAULT gst_rtp_ldac_pay_debug
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#define parent_class gst_rtp_ldac_pay_parent_class
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G_DEFINE_TYPE (GstRtpLdacPay, gst_rtp_ldac_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static GstStaticPadTemplate gst_rtp_ldac_pay_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-ldac, "
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"channels = (int) [ 1, 2 ], "
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"rate = (int) { 44100, 48000, 88200, 96000 }")
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);
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static GstStaticPadTemplate gst_rtp_ldac_pay_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) audio,"
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) { 44100, 48000, 88200, 96000 },"
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"encoding-name = (string) \"X-GST-LDAC\"")
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);
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static gboolean gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload *
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payload, GstBuffer * buffer);
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static void
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gst_rtp_ldac_pay_class_init (GstRtpLdacPayClass * klass)
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{
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GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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payload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_ldac_pay_set_caps);
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payload_class->handle_buffer =
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GST_DEBUG_FUNCPTR (gst_rtp_ldac_pay_handle_buffer);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_ldac_pay_sink_factory);
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gst_element_class_add_static_pad_template (element_class,
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&gst_rtp_ldac_pay_src_factory);
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gst_element_class_set_static_metadata (element_class, "RTP packet payloader",
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"Codec/Payloader/Network", "Payload LDAC audio as RTP packets",
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"Sanchayan Maity <sanchayan@asymptotic.io>");
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GST_DEBUG_CATEGORY_INIT (gst_rtp_ldac_pay_debug, "rtpldacpay", 0,
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"RTP LDAC payloader");
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}
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static void
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gst_rtp_ldac_pay_init (GstRtpLdacPay * self)
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{
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}
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static gboolean
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gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
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GstStructure *structure;
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gint rate;
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if (GST_RTP_BASE_PAYLOAD_MTU (ldacpay) < GST_LDAC_MTU_REQUIRED) {
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GST_ERROR_OBJECT (ldacpay, "Invalid MTU %d, should be >= %d",
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GST_RTP_BASE_PAYLOAD_MTU (ldacpay), GST_LDAC_MTU_REQUIRED);
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return FALSE;
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}
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "rate", &rate)) {
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GST_ERROR_OBJECT (ldacpay, "Failed to get audio rate from caps");
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return FALSE;
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}
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "X-GST-LDAC", rate);
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return gst_rtp_base_payload_set_outcaps (payload, NULL);
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}
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/*
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* LDAC encoder does not handle split frames. Currently, the encoder will
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* always emit 660 bytes worth of payload encapsulating multiple LDAC frames.
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* This is as per eqmid and GST_LDAC_MTU_REQUIRED passed for configuring the
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* encoder upstream. Since the encoder always emit full frames and we do not
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* need to handle frame splitting, we do not use an adapter and also push out
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* the buffer as it is received.
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*/
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static GstFlowReturn
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gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
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{
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GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
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GstBuffer *outbuf;
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GstClockTime outbuf_frame_duration, outbuf_pts;
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gsize buf_sz;
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
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(ldacpay), GST_RTP_HEADER_LENGTH, 0, 0);
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outbuf_pts = GST_BUFFER_PTS (buffer);
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outbuf_frame_duration = GST_BUFFER_DURATION (buffer);
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buf_sz = gst_buffer_get_size (buffer);
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gst_rtp_copy_audio_meta (ldacpay, outbuf, buffer);
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outbuf = gst_buffer_append (outbuf, buffer);
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GST_BUFFER_PTS (outbuf) = outbuf_pts;
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GST_BUFFER_DURATION (outbuf) = outbuf_frame_duration;
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GST_DEBUG_OBJECT (ldacpay,
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"Pushing %" G_GSIZE_FORMAT " bytes: %" GST_TIME_FORMAT, buf_sz,
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GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
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return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (ldacpay), outbuf);
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}
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gboolean
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gst_rtp_ldac_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpldacpay", GST_RANK_NONE,
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GST_TYPE_RTP_LDAC_PAY);
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}
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55
gst/rtp/gstrtpldacpay.h
Normal file
55
gst/rtp/gstrtpldacpay.h
Normal file
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@ -0,0 +1,55 @@
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/* GStreamer RTP LDAC payloader
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* Copyright (C) 2020 Asymptotic <sanchayan@asymptotic.io>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/rtp/gstrtpbasepayload.h>
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#include <gst/rtp/gstrtpbuffer.h>
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G_BEGIN_DECLS
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#define GST_TYPE_RTP_LDAC_PAY \
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(gst_rtp_ldac_pay_get_type())
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#define GST_RTP_LDAC_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_LDAC_PAY,\
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GstRtpLdacPay))
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#define GST_RTP_LDAC_PAY_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_LDAC_PAY,\
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GstRtpLdacPayClass))
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#define GST_IS_RTP_LDAC_PAY(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_LDAC_PAY))
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#define GST_IS_RTP_LDAC_PAY_CLASS(obj) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_LDAC_PAY))
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typedef struct _GstRtpLdacPay GstRtpLdacPay;
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typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass;
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struct _GstRtpLdacPay {
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GstRTPBasePayload base;
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};
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struct _GstRtpLdacPayClass {
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GstRTPBasePayloadClass parent_class;
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};
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GType gst_rtp_ldac_pay_get_type(void);
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gboolean gst_rtp_ldac_pay_plugin_init (GstPlugin * plugin);
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G_END_DECLS
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@ -61,6 +61,7 @@ rtp_sources = [
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'gstrtpL16pay.c',
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'gstrtpL16pay.c',
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'gstrtpL24depay.c',
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'gstrtpL24depay.c',
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'gstrtpL24pay.c',
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'gstrtpL24pay.c',
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'gstrtpldacpay.c',
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'gstasteriskh263.c',
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'gstasteriskh263.c',
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'gstrtpmp1sdepay.c',
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'gstrtpmp1sdepay.c',
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'gstrtpmp2tdepay.c',
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'gstrtpmp2tdepay.c',
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