--- /dev/null
+/* 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);
+}
--- /dev/null
+/* 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
projects / platform / upstream / gstreamer.git / commitdiff