commit: cf1b18cbd8da5490ace34d387679c9f9242dea34 Author: Igor V. Kovalenko <igor.v.kovalenko <AT> gmail <DOT> com> AuthorDate: Sat Feb 26 17:43:46 2022 +0000 Commit: Sam James <sam <AT> gentoo <DOT> org> CommitDate: Tue Mar 8 13:47:09 2022 +0000 URL: https://gitweb.gentoo.org/repo/gentoo.git/commit/?id=cf1b18cb
media-libs/gst-plugins-good: Backport ldac payload fix to 1.20.0 Signed-off-by: Igor V. Kovalenko <igor.v.kovalenko <AT> gmail.com> Signed-off-by: Sam James <sam <AT> gentoo.org> .../gst-plugins-good-1.20.0-ldac-rtp-header.patch | 163 +++++++++++++++++++++ .../gst-plugins-good-1.20.0.ebuild | 1 + 2 files changed, 164 insertions(+) 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 <[email protected]> +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() {
