Make a2dpsink to act like a bin and split the payloader.

1 files changed, 337 insertions, 0 deletions

diff --git a/audio/gstrtpsbcpay.c b/audio/gstrtpsbcpay.c
new file mode 100644
index 00000000..68aa28a9
--- /dev/null
+++ b/audio/gstrtpsbcpay.c
@@ -0,0 +1,337 @@
+/*
+ *
+ *  BlueZ - Bluetooth protocol stack for Linux
+ *
+ *  Copyright (C) 2004-2007  Marcel Holtmann <marcel@holtmann.org>
+ *
+ *
+ *  This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Lesser General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2.1 of the License, or (at your option) any later version.
+ *
+ *  This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Lesser General Public License for more details.
+ *
+ *  You should have received a copy of the GNU Lesser General Public
+ *  License along with this library; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstrtpsbcpay.h"
+#include <math.h>
+#include <string.h>
+
+#define RTP_SBC_PAYLOAD_HEADER_SIZE 1
+#define DEFAULT_MIN_FRAMES 0
+#define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE)
+
+#if __BYTE_ORDER == __LITTLE_ENDIAN
+
+struct rtp_payload {
+        guint8 frame_count:4;
+        guint8 rfa0:1;
+        guint8 is_last_fragment:1;
+        guint8 is_first_fragment:1;
+        guint8 is_fragmented:1;
+} __attribute__ ((packed));
+
+#elif __BYTE_ORDER == __BIG_ENDIAN
+
+struct rtp_payload {
+        guint8 is_fragmented:1;
+        guint8 is_first_fragment:1;
+        guint8 is_last_fragment:1;
+        guint8 rfa0:1;
+        guint8 frame_count:4;
+} __attribute__ ((packed));
+
+#else
+#error "Unknown byte order"
+#endif
+
+enum {
+	PROP_0,
+	PROP_MIN_FRAMES
+};
+
+GST_DEBUG_CATEGORY_STATIC(gst_rtp_sbc_pay_debug);
+#define GST_CAT_DEFAULT gst_rtp_sbc_pay_debug
+
+GST_BOILERPLATE(GstRtpSBCPay, gst_rtp_sbc_pay, GstBaseRTPPayload,
+		GST_TYPE_BASE_RTP_PAYLOAD);
+
+static const GstElementDetails gst_rtp_sbc_pay_details =
+	GST_ELEMENT_DETAILS("RTP packet payloader",
+				"Codec/Payloader/Network",
+				"Payload SBC audio as RTP packets",
+				"Thiago Sousa Santos "
+				"<thiagoss@lcc.ufcg.edu.br>");
+
+static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
+	GST_STATIC_PAD_TEMPLATE("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+		GST_STATIC_CAPS("audio/x-sbc, " /* FIXME remove those caps? */
+				"rate = (int) { 16000, 32000, 44100, 48000 }, "
+				"channels = (int) [ 1, 2 ], "
+				"mode = (string) { mono, dual, stereo, joint }, "
+				"blocks = (int) { 4, 8, 12, 16 }, "
+				"subbands = (int) { 4, 8 }, "
+				"allocation = (string) { snr, loudness },"
+				"bitpool = (int) [ 2, 64 ]; ")
+	);
+
+static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory =
+	GST_STATIC_PAD_TEMPLATE("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+		GST_STATIC_CAPS("application/x-rtp") /* FIXME put things here */
+	);
+
+static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
+                        const GValue * value, GParamSpec * pspec);
+static void gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id,
+                        GValue * value, GParamSpec * pspec);
+
+static gint gst_rtp_sbc_pay_get_frame_len(gint subbands, gint channels,
+		gint blocks, gint bitpool, const gchar* channel_mode)
+{
+	gint len;
+	gint join;
+
+	len = 4 + (4 * subbands * channels)/8;
+
+	if (strcmp(channel_mode, "mono") == 0 ||
+		strcmp(channel_mode, "dual") == 0)
+		len += ((blocks * channels * bitpool)+7) / 8;
+	else {
+		join = strcmp(channel_mode, "joint") == 0 ? 1 : 0;
+		len += ((join * subbands + blocks * bitpool)+7)/8;
+	}
+
+	return len;
+}
+
+static gboolean gst_rtp_sbc_pay_set_caps(GstBaseRTPPayload *payload,
+			GstCaps *caps)
+{
+	GstRtpSBCPay *sbcpay;
+	gint rate, subbands, channels, blocks, bitpool;
+	gint frame_len;
+	const gchar* channel_mode;
+	GstStructure *structure;
+
+	sbcpay = GST_RTP_SBC_PAY(payload);
+
+	structure = gst_caps_get_structure(caps, 0);
+	if (!gst_structure_get_int(structure, "rate", &rate))
+		return FALSE;
+	if (!gst_structure_get_int(structure, "channels", &channels))
+		return FALSE;
+	if (!gst_structure_get_int(structure, "blocks", &blocks))
+		return FALSE;
+	if (!gst_structure_get_int(structure, "bitpool", &bitpool))
+		return FALSE;
+	if (!gst_structure_get_int(structure, "subbands", &subbands))
+		return FALSE;
+
+	channel_mode = gst_structure_get_string(structure, "mode");
+	if (!channel_mode)
+		return FALSE;
+
+	frame_len = gst_rtp_sbc_pay_get_frame_len(subbands, channels, blocks,
+				bitpool, channel_mode);
+
+	sbcpay->frame_length = frame_len;
+
+	gst_basertppayload_set_options (payload, "audio", FALSE, "SBC", rate);
+
+	GST_DEBUG_OBJECT(payload, "calculated frame length: %d ", frame_len);
+
+	return gst_basertppayload_set_outcaps (payload, NULL);
+}
+
+static GstFlowReturn gst_rtp_sbc_pay_flush_buffers(GstRtpSBCPay *sbcpay)
+{
+	guint available;
+	guint max_payload;
+	GstBuffer* outbuf;
+	guint8 *payload_data;
+	guint8 *data;
+	struct rtp_payload *payload;
+
+	if (sbcpay->frame_length == 0) {
+		GST_ERROR_OBJECT(sbcpay, "Frame length is 0");
+		return GST_FLOW_ERROR;
+	}
+
+	available = gst_adapter_available(sbcpay->adapter);
+
+	max_payload = gst_rtp_buffer_calc_payload_len(
+		GST_BASE_RTP_PAYLOAD_MTU(sbcpay) - RTP_SBC_PAYLOAD_HEADER_SIZE,
+		0, 0);
+
+	max_payload = MIN(max_payload, available);
+
+	outbuf = gst_rtp_buffer_new_allocate(max_payload +
+			RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
+
+	gst_rtp_buffer_set_payload_type(outbuf,
+			GST_BASE_RTP_PAYLOAD_PT(sbcpay));
+
+	data = gst_adapter_take(sbcpay->adapter, max_payload);
+	payload_data = gst_rtp_buffer_get_payload(outbuf);
+
+	payload = (struct rtp_payload*) payload_data;
+	memset(payload, 0, sizeof(struct rtp_payload));
+	payload->frame_count = max_payload / sbcpay->frame_length;
+
+	memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data, max_payload);
+	g_free(data);
+
+	/* FIXME - timestamp it! */
+	GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", max_payload);
+
+	return gst_basertppayload_push (GST_BASE_RTP_PAYLOAD(sbcpay), outbuf);
+}
+
+static GstFlowReturn gst_rtp_sbc_pay_handle_buffer(GstBaseRTPPayload *payload,
+			GstBuffer *buffer)
+{
+	GstRtpSBCPay *sbcpay;
+	guint available;
+
+	sbcpay = GST_RTP_SBC_PAY(payload);
+	gst_adapter_push(sbcpay->adapter, gst_buffer_copy(buffer));
+
+	available = gst_adapter_available(sbcpay->adapter);
+	if (available + RTP_SBC_HEADER_TOTAL >=
+			GST_BASE_RTP_PAYLOAD_MTU(sbcpay) ||
+			(sbcpay->min_frames != -1 && available >
+			(sbcpay->min_frames * sbcpay->frame_length)))
+		return gst_rtp_sbc_pay_flush_buffers(sbcpay);
+
+	return GST_FLOW_OK;
+}
+
+static gboolean gst_rtp_sbc_pay_handle_event(GstPad *pad,
+				GstEvent *event)
+{
+	GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY(GST_PAD_PARENT(pad));
+
+	switch (GST_EVENT_TYPE(event)) {
+	case GST_EVENT_EOS:
+		gst_rtp_sbc_pay_flush_buffers(sbcpay);
+		break;
+	default:
+		break;
+	}
+
+	return FALSE;
+}
+
+static void gst_rtp_sbc_pay_base_init(gpointer g_class)
+{
+	GstElementClass *element_class = GST_ELEMENT_CLASS(g_class);
+
+	gst_element_class_add_pad_template(element_class,
+		gst_static_pad_template_get(&gst_rtp_sbc_pay_sink_factory));
+	gst_element_class_add_pad_template(element_class,
+		gst_static_pad_template_get(&gst_rtp_sbc_pay_src_factory));
+
+	gst_element_class_set_details(element_class, &gst_rtp_sbc_pay_details);
+}
+
+static void gst_rtp_sbc_pay_finalize(GObject *object)
+{
+	GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY(object);
+	g_object_unref (sbcpay->adapter);
+
+	GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
+}
+
+static void gst_rtp_sbc_pay_class_init(GstRtpSBCPayClass *klass)
+{
+	GObjectClass *gobject_class;
+	GstBaseRTPPayloadClass *payload_class =
+		GST_BASE_RTP_PAYLOAD_CLASS(klass);
+
+	gobject_class = G_OBJECT_CLASS(klass);
+	parent_class = g_type_class_peek_parent(klass);
+
+	gobject_class->finalize = GST_DEBUG_FUNCPTR(gst_rtp_sbc_pay_finalize);
+	gobject_class->set_property = GST_DEBUG_FUNCPTR(
+			gst_rtp_sbc_pay_set_property);
+	gobject_class->get_property = GST_DEBUG_FUNCPTR(
+			gst_rtp_sbc_pay_get_property);
+
+	payload_class->set_caps = GST_DEBUG_FUNCPTR(gst_rtp_sbc_pay_set_caps);
+	payload_class->handle_buffer = GST_DEBUG_FUNCPTR(
+			gst_rtp_sbc_pay_handle_buffer);
+	payload_class->handle_event = GST_DEBUG_FUNCPTR(
+			gst_rtp_sbc_pay_handle_event);
+
+	/* properties */
+	g_object_class_install_property (G_OBJECT_CLASS (klass),
+		PROP_MIN_FRAMES,
+		g_param_spec_int ("min-frames", "minimum frame number",
+		"Minimum quantity of frames to send in one packet "
+		"(-1 for maximum allowed by the mtu)",
+		-1, G_MAXINT, DEFAULT_MIN_FRAMES, G_PARAM_READWRITE));
+
+	GST_DEBUG_CATEGORY_INIT(gst_rtp_sbc_pay_debug, "rtpsbcpay", 0,
+				"RTP SBC payloader");
+}
+
+static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
+			const GValue * value, GParamSpec * pspec)
+{
+	GstRtpSBCPay *sbcpay;
+
+	sbcpay = GST_RTP_SBC_PAY (object);
+
+	switch (prop_id) {
+	case PROP_MIN_FRAMES:
+		sbcpay->min_frames = g_value_get_int(value);
+		break;
+	default:
+		G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+	break;
+	}
+}
+
+static void gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id,
+                        GValue * value, GParamSpec * pspec)
+{
+	GstRtpSBCPay *sbcpay;
+
+	sbcpay = GST_RTP_SBC_PAY (object);
+
+	switch (prop_id) {
+	case PROP_MIN_FRAMES:
+		g_value_set_int(value, sbcpay->min_frames);
+		break;
+	default:
+		G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+	break;
+	}
+}
+
+static void gst_rtp_sbc_pay_init(GstRtpSBCPay *self, GstRtpSBCPayClass *klass)
+{
+	self->adapter = gst_adapter_new();
+	self->frame_length = 0;
+
+	self->min_frames = DEFAULT_MIN_FRAMES;
+}
+
+gboolean gst_rtp_sbc_pay_plugin_init (GstPlugin * plugin)
+{
+	return gst_element_register (plugin, "rtpsbcpay",
+			GST_RANK_NONE, GST_TYPE_RTP_SBC_PAY);
+}
+