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authorMichael Mair-Keimberger <mmk@levelnine.at>2022-03-18 07:21:00 +0100
committerConrad Kostecki <conikost@gentoo.org>2022-03-28 00:07:07 +0200
commiteeeda2ee3785c457be61b74887e6dd84ccbe07c2 (patch)
tree3ebc98b85d4b13d046a95cdd741463800ae49a37
parentmedia-video/dvdrip: remove unused patch (diff)
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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>
-rw-r--r--media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch23
-rw-r--r--media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch163
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 c431b1fb6bd..00000000000
--- 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 f1fc4601a23..00000000000
--- 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
-