gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.c
Sanchayan Maity cc3419daf6 rtp: ldac: Set frame count information in payload
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.

Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 21:09:57 +05:30

228 lines
7.5 KiB
C

/* 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 "gstrtpelements.h"
#include "gstrtpldacpay.h"
#include "gstrtputils.h"
#define GST_RTP_LDAC_PAYLOAD_HEADER_SIZE 1
/* 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 ], "
"eqmid = (int) { 0, 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);
/**
* gst_rtp_ldac_pay_get_num_frames
* @eqmid: Encode Quality Mode Index
* @channels: Number of channels
*
* Returns: Number of LDAC frames per packet.
*/
static guint8
gst_rtp_ldac_pay_get_num_frames (gint eqmid, gint channels)
{
g_assert (channels == 1 || channels == 2);
switch (eqmid) {
/* Encode setting for High Quality */
case 0:
return 4 / channels;
/* Encode setting for Standard Quality */
case 1:
return 6 / channels;
/* Encode setting for Mobile use Quality */
case 2:
return 12 / channels;
default:
break;
}
g_assert_not_reached ();
/* If assertion gets compiled out */
return 6 / channels;
}
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 channels, eqmid, 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;
}
if (!gst_structure_get_int (structure, "channels", &channels)) {
GST_ERROR_OBJECT (ldacpay, "Failed to get audio rate from caps");
return FALSE;
}
if (!gst_structure_get_int (structure, "eqmid", &eqmid)) {
GST_ERROR_OBJECT (ldacpay, "Failed to get eqmid from caps");
return FALSE;
}
ldacpay->frame_count = gst_rtp_ldac_pay_get_num_frames (eqmid, channels);
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)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
GstBuffer *outbuf;
GstClockTime outbuf_frame_duration, outbuf_pts;
guint8 *payload_data;
gsize buf_sz;
outbuf =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(ldacpay), GST_RTP_LDAC_PAYLOAD_HEADER_SIZE, 0, 0);
/* Get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* Write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
/* Upper 3 fragment bits not used, ref A2DP v13, 4.3.4 */
payload_data[0] = ldacpay->frame_count & 0x0f;
gst_rtp_buffer_unmap (&rtp);
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);
}