media-libs/gst-plugins-good: remove unused patch(es)

Closes: https://github.com/gentoo/gentoo/pull/24633
Package-Manager: Portage-3.0.30, Repoman-3.0.3
Signed-off-by: Michael Mair-Keimberger <mmk@levelnine.at>
Signed-off-by: Conrad Kostecki <conikost@gentoo.org>

2 files changed, 0 insertions, 186 deletions

diff --git a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch b/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch
deleted file mode 100644
index c431b1fb6bd7..000000000000
--- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch
+++ /dev/null
@@ -1,23 +0,0 @@
-commit d03971dac7b32a6ffcbf161853e017f65ae7c22f
-Author: Heiko Becker <heirecka@exherbo.org>
-Date:   Fri Feb 11 21:35:54 2022 +0100
-
-    meson: Don't build lame plugin with -Dlame=disabled
-    
-    Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1686>
-
-diff --git a/ext/lame/meson.build b/ext/lame/meson.build
-index 2169fde6f4..3290f17e1e 100644
---- a/ext/lame/meson.build
-+++ b/ext/lame/meson.build
-@@ -1,5 +1,10 @@
-+lame_dep = dependency('', required: false)
- lame_option = get_option('lame')
- 
-+if lame_option.disabled()
-+  subdir_done()
-+endif
-+
- lame_extra_c_args = []
- lame_dep = cc.find_library('mp3lame', required: false)
- have_lame = cc.has_header_symbol('lame/lame.h', 'lame_init')
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
deleted file mode 100644
index f1fc4601a23a..000000000000
--- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch
+++ /dev/null
@@ -1,163 +0,0 @@
-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
-