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From cc3419daf60159394cd310c2405a78775b3f84db Mon Sep 17 00:00:00 2001
From: Sanchayan Maity <sanchayan@asymptotic.io>
Date: Thu, 24 Feb 2022 20:28:23 +0530
Subject: [PATCH] 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>
---
.../docs/gst_plugins_cache.json | 2 +-
.../gst-plugins-good/gst/rtp/gstrtpldacpay.c | 63 ++++++++++++++++++-
.../gst-plugins-good/gst/rtp/gstrtpldacpay.h | 1 +
3 files changed, 62 insertions(+), 4 deletions(-)
diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json
index 88bff47243..003546d59d 100644
--- a/docs/gst_plugins_cache.json
+++ b/docs/gst_plugins_cache.json
@@ -14678,7 +14678,7 @@
"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",
+ "caps": "audio/x-ldac:\n channels: [ 1, 2 ]\n eqmid: { (int)0, (int)1, (int)2 }\n rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n",
"direction": "sink",
"presence": "always"
},
diff --git a/gst/rtp/gstrtpldacpay.c b/gst/rtp/gstrtpldacpay.c
index 2b14b746fe..aa30673e7e 100644
--- a/gst/rtp/gstrtpldacpay.c
+++ b/gst/rtp/gstrtpldacpay.c
@@ -48,7 +48,7 @@
#include "gstrtpldacpay.h"
#include "gstrtputils.h"
-#define GST_RTP_HEADER_LENGTH 12
+#define GST_RTP_LDAC_PAYLOAD_HEADER_SIZE 1
/* MTU size required for LDAC A2DP streaming */
#define GST_LDAC_MTU_REQUIRED 679
@@ -64,6 +64,7 @@ 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 }")
);
@@ -81,6 +82,38 @@ static gboolean gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload,
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)
{
@@ -115,7 +148,7 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
GstStructure *structure;
- gint rate;
+ 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",
@@ -129,6 +162,18 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * 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);
@@ -145,14 +190,26 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
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_HEADER_LENGTH, 0, 0);
+ (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);
diff --git a/gst/rtp/gstrtpldacpay.h b/gst/rtp/gstrtpldacpay.h
index 0865ce7ade..0134491752 100644
--- a/gst/rtp/gstrtpldacpay.h
+++ b/gst/rtp/gstrtpldacpay.h
@@ -42,6 +42,7 @@ typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass;
struct _GstRtpLdacPay {
GstRTPBasePayload base;
+ guint8 frame_count;
};
struct _GstRtpLdacPayClass {
--
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