Gentoo Archives: gentoo-commits

From: Conrad Kostecki <conikost@g.o>
To: gentoo-commits@l.g.o
Subject: [gentoo-commits] repo/gentoo:master commit in: media-libs/gst-plugins-good/files/
Date: Sun, 27 Mar 2022 22:08:32
Message-Id: 1648418827.eeeda2ee3785c457be61b74887e6dd84ccbe07c2.conikost@gentoo
1 commit: eeeda2ee3785c457be61b74887e6dd84ccbe07c2
2 Author: Michael Mair-Keimberger <mmk <AT> levelnine <DOT> at>
3 AuthorDate: Fri Mar 18 06:21:00 2022 +0000
4 Commit: Conrad Kostecki <conikost <AT> gentoo <DOT> org>
5 CommitDate: Sun Mar 27 22:07:07 2022 +0000
6 URL: https://gitweb.gentoo.org/repo/gentoo.git/commit/?id=eeeda2ee
7
8 media-libs/gst-plugins-good: remove unused patch(es)
9
10 Closes: https://github.com/gentoo/gentoo/pull/24633
11 Package-Manager: Portage-3.0.30, Repoman-3.0.3
12 Signed-off-by: Michael Mair-Keimberger <mmk <AT> levelnine.at>
13 Signed-off-by: Conrad Kostecki <conikost <AT> gentoo.org>
14
15 ...t-plugins-good-1.20.0-lame-feature-option.patch | 23 ---
16 .../gst-plugins-good-1.20.0-ldac-rtp-header.patch | 163 ---------------------
17 2 files changed, 186 deletions(-)
18
19 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
20 deleted file mode 100644
21 index c431b1fb6bd7..000000000000
22 --- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch
23 +++ /dev/null
24 @@ -1,23 +0,0 @@
25 -commit d03971dac7b32a6ffcbf161853e017f65ae7c22f
26 -Author: Heiko Becker <heirecka@×××××××.org>
27 -Date: Fri Feb 11 21:35:54 2022 +0100
28 -
29 - meson: Don't build lame plugin with -Dlame=disabled
30 -
31 - Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1686>
32 -
33 -diff --git a/ext/lame/meson.build b/ext/lame/meson.build
34 -index 2169fde6f4..3290f17e1e 100644
35 ---- a/ext/lame/meson.build
36 -+++ b/ext/lame/meson.build
37 -@@ -1,5 +1,10 @@
38 -+lame_dep = dependency('', required: false)
39 - lame_option = get_option('lame')
40 -
41 -+if lame_option.disabled()
42 -+ subdir_done()
43 -+endif
44 -+
45 - lame_extra_c_args = []
46 - lame_dep = cc.find_library('mp3lame', required: false)
47 - have_lame = cc.has_header_symbol('lame/lame.h', 'lame_init')
48
49 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
50 deleted file mode 100644
51 index f1fc4601a23a..000000000000
52 --- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch
53 +++ /dev/null
54 @@ -1,163 +0,0 @@
55 -From cc3419daf60159394cd310c2405a78775b3f84db Mon Sep 17 00:00:00 2001
56 -From: Sanchayan Maity <sanchayan@××××××××××.io>
57 -Date: Thu, 24 Feb 2022 20:28:23 +0530
58 -Subject: [PATCH] rtp: ldac: Set frame count information in payload
59 -
60 -The RTP payload seems to be required as it carries the frame count
61 -information. Also, gst_rtp_base_payload_allocate_output_buffer had
62 -the second argument incorrect.
63 -
64 -Strangely some devices like Shanling MP4 and Sony XM3 would still
65 -work without this while some like the Sony XM4 do not.
66 -
67 -Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
68 ----
69 - .../docs/gst_plugins_cache.json | 2 +-
70 - .../gst-plugins-good/gst/rtp/gstrtpldacpay.c | 63 ++++++++++++++++++-
71 - .../gst-plugins-good/gst/rtp/gstrtpldacpay.h | 1 +
72 - 3 files changed, 62 insertions(+), 4 deletions(-)
73 -
74 -diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json
75 -index 88bff47243..003546d59d 100644
76 ---- a/docs/gst_plugins_cache.json
77 -+++ b/docs/gst_plugins_cache.json
78 -@@ -14678,7 +14678,7 @@
79 - "long-name": "RTP packet payloader",
80 - "pad-templates": {
81 - "sink": {
82 -- "caps": "audio/x-ldac:\n channels: [ 1, 2 ]\n rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n",
83 -+ "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",
84 - "direction": "sink",
85 - "presence": "always"
86 - },
87 -diff --git a/gst/rtp/gstrtpldacpay.c b/gst/rtp/gstrtpldacpay.c
88 -index 2b14b746fe..aa30673e7e 100644
89 ---- a/gst/rtp/gstrtpldacpay.c
90 -+++ b/gst/rtp/gstrtpldacpay.c
91 -@@ -48,7 +48,7 @@
92 - #include "gstrtpldacpay.h"
93 - #include "gstrtputils.h"
94 -
95 --#define GST_RTP_HEADER_LENGTH 12
96 -+#define GST_RTP_LDAC_PAYLOAD_HEADER_SIZE 1
97 - /* MTU size required for LDAC A2DP streaming */
98 - #define GST_LDAC_MTU_REQUIRED 679
99 -
100 -@@ -64,6 +64,7 @@ static GstStaticPadTemplate gst_rtp_ldac_pay_sink_factory =
101 - GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
102 - GST_STATIC_CAPS ("audio/x-ldac, "
103 - "channels = (int) [ 1, 2 ], "
104 -+ "eqmid = (int) { 0, 1, 2 }, "
105 - "rate = (int) { 44100, 48000, 88200, 96000 }")
106 - );
107 -
108 -@@ -81,6 +82,38 @@ static gboolean gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload,
109 - static GstFlowReturn gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload *
110 - payload, GstBuffer * buffer);
111 -
112 -+/**
113 -+ * gst_rtp_ldac_pay_get_num_frames
114 -+ * @eqmid: Encode Quality Mode Index
115 -+ * @channels: Number of channels
116 -+ *
117 -+ * Returns: Number of LDAC frames per packet.
118 -+ */
119 -+static guint8
120 -+gst_rtp_ldac_pay_get_num_frames (gint eqmid, gint channels)
121 -+{
122 -+ g_assert (channels == 1 || channels == 2);
123 -+
124 -+ switch (eqmid) {
125 -+ /* Encode setting for High Quality */
126 -+ case 0:
127 -+ return 4 / channels;
128 -+ /* Encode setting for Standard Quality */
129 -+ case 1:
130 -+ return 6 / channels;
131 -+ /* Encode setting for Mobile use Quality */
132 -+ case 2:
133 -+ return 12 / channels;
134 -+ default:
135 -+ break;
136 -+ }
137 -+
138 -+ g_assert_not_reached ();
139 -+
140 -+ /* If assertion gets compiled out */
141 -+ return 6 / channels;
142 -+}
143 -+
144 - static void
145 - gst_rtp_ldac_pay_class_init (GstRtpLdacPayClass * klass)
146 - {
147 -@@ -115,7 +148,7 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
148 - {
149 - GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
150 - GstStructure *structure;
151 -- gint rate;
152 -+ gint channels, eqmid, rate;
153 -
154 - if (GST_RTP_BASE_PAYLOAD_MTU (ldacpay) < GST_LDAC_MTU_REQUIRED) {
155 - GST_ERROR_OBJECT (ldacpay, "Invalid MTU %d, should be >= %d",
156 -@@ -129,6 +162,18 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
157 - return FALSE;
158 - }
159 -
160 -+ if (!gst_structure_get_int (structure, "channels", &channels)) {
161 -+ GST_ERROR_OBJECT (ldacpay, "Failed to get audio rate from caps");
162 -+ return FALSE;
163 -+ }
164 -+
165 -+ if (!gst_structure_get_int (structure, "eqmid", &eqmid)) {
166 -+ GST_ERROR_OBJECT (ldacpay, "Failed to get eqmid from caps");
167 -+ return FALSE;
168 -+ }
169 -+
170 -+ ldacpay->frame_count = gst_rtp_ldac_pay_get_num_frames (eqmid, channels);
171 -+
172 - gst_rtp_base_payload_set_options (payload, "audio", TRUE, "X-GST-LDAC", rate);
173 -
174 - return gst_rtp_base_payload_set_outcaps (payload, NULL);
175 -@@ -145,14 +190,26 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
176 - static GstFlowReturn
177 - gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
178 - {
179 -+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
180 - GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload);
181 - GstBuffer *outbuf;
182 - GstClockTime outbuf_frame_duration, outbuf_pts;
183 -+ guint8 *payload_data;
184 - gsize buf_sz;
185 -
186 - outbuf =
187 - gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
188 -- (ldacpay), GST_RTP_HEADER_LENGTH, 0, 0);
189 -+ (ldacpay), GST_RTP_LDAC_PAYLOAD_HEADER_SIZE, 0, 0);
190 -+
191 -+ /* Get payload */
192 -+ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
193 -+
194 -+ /* Write header and copy data into payload */
195 -+ payload_data = gst_rtp_buffer_get_payload (&rtp);
196 -+ /* Upper 3 fragment bits not used, ref A2DP v13, 4.3.4 */
197 -+ payload_data[0] = ldacpay->frame_count & 0x0f;
198 -+
199 -+ gst_rtp_buffer_unmap (&rtp);
200 -
201 - outbuf_pts = GST_BUFFER_PTS (buffer);
202 - outbuf_frame_duration = GST_BUFFER_DURATION (buffer);
203 -diff --git a/gst/rtp/gstrtpldacpay.h b/gst/rtp/gstrtpldacpay.h
204 -index 0865ce7ade..0134491752 100644
205 ---- a/gst/rtp/gstrtpldacpay.h
206 -+++ b/gst/rtp/gstrtpldacpay.h
207 -@@ -42,6 +42,7 @@ typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass;
208 -
209 - struct _GstRtpLdacPay {
210 - GstRTPBasePayload base;
211 -+ guint8 frame_count;
212 - };
213 -
214 - struct _GstRtpLdacPayClass {
215 ---
216 -GitLab
217 -