media-libs/gst-plugins-good: Backport ldac payload fix to 1.20.0

Signed-off-by: Igor V. Kovalenko <igor.v.kovalenko@gmail.com>
Signed-off-by: Sam James <sam@gentoo.org>

2 files changed, 164 insertions, 0 deletions

diff --git a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch b/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch
new file mode 100644
index 000000000000..f1fc4601a23a
--- /dev/null
+++ b/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch
@@ -0,0 +1,163 @@
+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 {
+-- 
+GitLab
+
diff --git a/media-libs/gst-plugins-good/gst-plugins-good-1.20.0.ebuild b/media-libs/gst-plugins-good/gst-plugins-good-1.20.0.ebuild
index 82182c710194..d696e4023d0c 100644
--- a/media-libs/gst-plugins-good/gst-plugins-good-1.20.0.ebuild
+++ b/media-libs/gst-plugins-good/gst-plugins-good-1.20.0.ebuild
@@ -27,6 +27,7 @@ DOCS=( AUTHORS ChangeLog NEWS README RELEASE )
 # Fixes backported to 1.20.0, to be removed in 1.20.1+
 PATCHES=(
 	"${FILESDIR}/gst-plugins-good-1.20.0-lame-feature-option.patch"
+	"${FILESDIR}/gst-plugins-good-1.20.0-ldac-rtp-header.patch"
 )
 
 multilib_src_configure() {