diff options
Diffstat (limited to 'media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch')
| -rw-r--r-- | media-libs/gst-plugins-good/files/gst-plugins-good-1.20.0-ldac-rtp-header.patch | 163 |
1 files changed, 0 insertions, 163 deletions
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 - |