/* GStreamer RTP LDAC payloader * Copyright (C) 2020 Asymptotic * * 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 #endif #include #include "gstrtpelements.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); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpldacpay, "rtpldacpay", GST_RANK_NONE, GST_TYPE_RTP_LDAC_PAY, rtp_element_init (plugin)); 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 "); 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); }