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commit: eeeda2ee3785c457be61b74887e6dd84ccbe07c2 |
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Author: Michael Mair-Keimberger <mmk <AT> levelnine <DOT> at> |
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AuthorDate: Fri Mar 18 06:21:00 2022 +0000 |
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Commit: Conrad Kostecki <conikost <AT> gentoo <DOT> org> |
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CommitDate: Sun Mar 27 22:07:07 2022 +0000 |
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URL: https://gitweb.gentoo.org/repo/gentoo.git/commit/?id=eeeda2ee |
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|
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media-libs/gst-plugins-good: remove unused patch(es) |
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|
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Closes: https://github.com/gentoo/gentoo/pull/24633 |
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Package-Manager: Portage-3.0.30, Repoman-3.0.3 |
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Signed-off-by: Michael Mair-Keimberger <mmk <AT> levelnine.at> |
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Signed-off-by: Conrad Kostecki <conikost <AT> gentoo.org> |
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|
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...t-plugins-good-1.20.0-lame-feature-option.patch | 23 --- |
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.../gst-plugins-good-1.20.0-ldac-rtp-header.patch | 163 --------------------- |
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2 files changed, 186 deletions(-) |
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|
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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 |
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deleted file mode 100644 |
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index c431b1fb6bd7..000000000000 |
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--- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-lame-feature-option.patch |
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+++ /dev/null |
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@@ -1,23 +0,0 @@ |
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-commit d03971dac7b32a6ffcbf161853e017f65ae7c22f |
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-Author: Heiko Becker <heirecka@×××××××.org> |
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-Date: Fri Feb 11 21:35:54 2022 +0100 |
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- |
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- meson: Don't build lame plugin with -Dlame=disabled |
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- |
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- Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1686> |
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- |
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-diff --git a/ext/lame/meson.build b/ext/lame/meson.build |
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-index 2169fde6f4..3290f17e1e 100644 |
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---- a/ext/lame/meson.build |
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-+++ b/ext/lame/meson.build |
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-@@ -1,5 +1,10 @@ |
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-+lame_dep = dependency('', required: false) |
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- lame_option = get_option('lame') |
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- |
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-+if lame_option.disabled() |
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-+ subdir_done() |
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-+endif |
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-+ |
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- lame_extra_c_args = [] |
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- lame_dep = cc.find_library('mp3lame', required: false) |
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- have_lame = cc.has_header_symbol('lame/lame.h', 'lame_init') |
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|
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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 |
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deleted file mode 100644 |
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index f1fc4601a23a..000000000000 |
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--- a/media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch |
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+++ /dev/null |
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@@ -1,163 +0,0 @@ |
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-From cc3419daf60159394cd310c2405a78775b3f84db Mon Sep 17 00:00:00 2001 |
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-From: Sanchayan Maity <sanchayan@××××××××××.io> |
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-Date: Thu, 24 Feb 2022 20:28:23 +0530 |
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-Subject: [PATCH] rtp: ldac: Set frame count information in payload |
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- |
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-The RTP payload seems to be required as it carries the frame count |
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-information. Also, gst_rtp_base_payload_allocate_output_buffer had |
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-the second argument incorrect. |
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- |
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-Strangely some devices like Shanling MP4 and Sony XM3 would still |
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-work without this while some like the Sony XM4 do not. |
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- |
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-Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797> |
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---- |
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- .../docs/gst_plugins_cache.json | 2 +- |
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- .../gst-plugins-good/gst/rtp/gstrtpldacpay.c | 63 ++++++++++++++++++- |
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- .../gst-plugins-good/gst/rtp/gstrtpldacpay.h | 1 + |
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- 3 files changed, 62 insertions(+), 4 deletions(-) |
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- |
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-diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json |
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-index 88bff47243..003546d59d 100644 |
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---- a/docs/gst_plugins_cache.json |
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-+++ b/docs/gst_plugins_cache.json |
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-@@ -14678,7 +14678,7 @@ |
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- "long-name": "RTP packet payloader", |
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- "pad-templates": { |
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- "sink": { |
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-- "caps": "audio/x-ldac:\n channels: [ 1, 2 ]\n rate: { (int)44100, (int)48000, (int)88200, (int)96000 }\n", |
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-+ "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", |
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- "direction": "sink", |
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- "presence": "always" |
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- }, |
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-diff --git a/gst/rtp/gstrtpldacpay.c b/gst/rtp/gstrtpldacpay.c |
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-index 2b14b746fe..aa30673e7e 100644 |
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---- a/gst/rtp/gstrtpldacpay.c |
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-+++ b/gst/rtp/gstrtpldacpay.c |
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-@@ -48,7 +48,7 @@ |
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- #include "gstrtpldacpay.h" |
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- #include "gstrtputils.h" |
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- |
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--#define GST_RTP_HEADER_LENGTH 12 |
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-+#define GST_RTP_LDAC_PAYLOAD_HEADER_SIZE 1 |
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- /* MTU size required for LDAC A2DP streaming */ |
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- #define GST_LDAC_MTU_REQUIRED 679 |
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- |
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-@@ -64,6 +64,7 @@ static GstStaticPadTemplate gst_rtp_ldac_pay_sink_factory = |
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- GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, |
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- GST_STATIC_CAPS ("audio/x-ldac, " |
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- "channels = (int) [ 1, 2 ], " |
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-+ "eqmid = (int) { 0, 1, 2 }, " |
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- "rate = (int) { 44100, 48000, 88200, 96000 }") |
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- ); |
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- |
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-@@ -81,6 +82,38 @@ static gboolean gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, |
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- static GstFlowReturn gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload * |
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- payload, GstBuffer * buffer); |
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- |
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-+/** |
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-+ * gst_rtp_ldac_pay_get_num_frames |
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-+ * @eqmid: Encode Quality Mode Index |
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-+ * @channels: Number of channels |
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-+ * |
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-+ * Returns: Number of LDAC frames per packet. |
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-+ */ |
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-+static guint8 |
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-+gst_rtp_ldac_pay_get_num_frames (gint eqmid, gint channels) |
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-+{ |
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-+ g_assert (channels == 1 || channels == 2); |
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-+ |
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-+ switch (eqmid) { |
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-+ /* Encode setting for High Quality */ |
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-+ case 0: |
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-+ return 4 / channels; |
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-+ /* Encode setting for Standard Quality */ |
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-+ case 1: |
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-+ return 6 / channels; |
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-+ /* Encode setting for Mobile use Quality */ |
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-+ case 2: |
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-+ return 12 / channels; |
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-+ default: |
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-+ break; |
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-+ } |
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-+ |
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-+ g_assert_not_reached (); |
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-+ |
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-+ /* If assertion gets compiled out */ |
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-+ return 6 / channels; |
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-+} |
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-+ |
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- static void |
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- gst_rtp_ldac_pay_class_init (GstRtpLdacPayClass * klass) |
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- { |
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-@@ -115,7 +148,7 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps) |
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- { |
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- GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload); |
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- GstStructure *structure; |
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-- gint rate; |
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-+ gint channels, eqmid, rate; |
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- |
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- if (GST_RTP_BASE_PAYLOAD_MTU (ldacpay) < GST_LDAC_MTU_REQUIRED) { |
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- GST_ERROR_OBJECT (ldacpay, "Invalid MTU %d, should be >= %d", |
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-@@ -129,6 +162,18 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps) |
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- return FALSE; |
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- } |
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- |
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-+ if (!gst_structure_get_int (structure, "channels", &channels)) { |
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-+ GST_ERROR_OBJECT (ldacpay, "Failed to get audio rate from caps"); |
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-+ return FALSE; |
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-+ } |
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-+ |
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-+ if (!gst_structure_get_int (structure, "eqmid", &eqmid)) { |
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-+ GST_ERROR_OBJECT (ldacpay, "Failed to get eqmid from caps"); |
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-+ return FALSE; |
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-+ } |
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-+ |
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-+ ldacpay->frame_count = gst_rtp_ldac_pay_get_num_frames (eqmid, channels); |
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-+ |
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- gst_rtp_base_payload_set_options (payload, "audio", TRUE, "X-GST-LDAC", rate); |
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- |
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- return gst_rtp_base_payload_set_outcaps (payload, NULL); |
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-@@ -145,14 +190,26 @@ gst_rtp_ldac_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps) |
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- static GstFlowReturn |
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- gst_rtp_ldac_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer) |
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- { |
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-+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; |
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- GstRtpLdacPay *ldacpay = GST_RTP_LDAC_PAY (payload); |
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- GstBuffer *outbuf; |
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- GstClockTime outbuf_frame_duration, outbuf_pts; |
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-+ guint8 *payload_data; |
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- gsize buf_sz; |
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- |
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- outbuf = |
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- gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD |
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-- (ldacpay), GST_RTP_HEADER_LENGTH, 0, 0); |
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-+ (ldacpay), GST_RTP_LDAC_PAYLOAD_HEADER_SIZE, 0, 0); |
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-+ |
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-+ /* Get payload */ |
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-+ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); |
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-+ |
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-+ /* Write header and copy data into payload */ |
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-+ payload_data = gst_rtp_buffer_get_payload (&rtp); |
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-+ /* Upper 3 fragment bits not used, ref A2DP v13, 4.3.4 */ |
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-+ payload_data[0] = ldacpay->frame_count & 0x0f; |
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-+ |
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-+ gst_rtp_buffer_unmap (&rtp); |
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- |
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- outbuf_pts = GST_BUFFER_PTS (buffer); |
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- outbuf_frame_duration = GST_BUFFER_DURATION (buffer); |
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-diff --git a/gst/rtp/gstrtpldacpay.h b/gst/rtp/gstrtpldacpay.h |
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-index 0865ce7ade..0134491752 100644 |
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---- a/gst/rtp/gstrtpldacpay.h |
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-+++ b/gst/rtp/gstrtpldacpay.h |
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-@@ -42,6 +42,7 @@ typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass; |
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- |
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- struct _GstRtpLdacPay { |
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- GstRTPBasePayload base; |
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-+ guint8 frame_count; |
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- }; |
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- |
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- struct _GstRtpLdacPayClass { |
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--- |
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-GitLab |
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- |