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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/1804>
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3 changed files with 62 additions and 4 deletions
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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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@ -48,7 +48,7 @@
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#include "gstrtpldacpay.h"
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#include "gstrtputils.h"
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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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@ -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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@ -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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* 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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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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g_assert_not_reached ();
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/* If assertion gets compiled out */
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return 6 / channels;
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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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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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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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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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ldacpay->frame_count = gst_rtp_ldac_pay_get_num_frames (eqmid, channels);
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "X-GST-LDAC", rate);
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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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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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/* Get payload */
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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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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gst_rtp_buffer_unmap (&rtp);
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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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@ -42,6 +42,7 @@ typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass;
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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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struct _GstRtpLdacPayClass {
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