From cc3419daf60159394cd310c2405a78775b3f84db Mon Sep 17 00:00:00 2001 From: Sanchayan Maity 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: --- .../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/subprojects/gst-plugins-good/docs/gst_plugins_cache.json b/subprojects/gst-plugins-good/docs/gst_plugins_cache.json index 88bff47243..003546d59d 100644 --- a/subprojects/gst-plugins-good/docs/gst_plugins_cache.json +++ b/subprojects/gst-plugins-good/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/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.c b/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.c index 2b14b746fe..aa30673e7e 100644 --- a/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.c +++ b/subprojects/gst-plugins-good/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/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.h b/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.h index 0865ce7ade..0134491752 100644 --- a/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.h +++ b/subprojects/gst-plugins-good/gst/rtp/gstrtpldacpay.h @@ -42,6 +42,7 @@ typedef struct _GstRtpLdacPayClass GstRtpLdacPayClass; struct _GstRtpLdacPay { GstRTPBasePayload base; + guint8 frame_count; }; struct _GstRtpLdacPayClass {