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path: root/audio/gstrtpsbcpay.c
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/*
 *
 *  BlueZ - Bluetooth protocol stack for Linux
 *
 *  Copyright (C) 2004-2008  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, "
				"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, "
			"media = (string) \"audio\","
			"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
			"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
			"encoding-name = (string) \"SBC\"")
	);

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", TRUE, "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;
	guint frame_count;
	guint payload_length;
	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);
	frame_count = max_payload / sbcpay->frame_length;
	payload_length = frame_count * sbcpay->frame_length;
	if (payload_length == 0) /* Nothing to send */
		return GST_FLOW_OK;

	outbuf = gst_rtp_buffer_new_allocate(payload_length +
			RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);

	gst_rtp_buffer_set_payload_type(outbuf,
			GST_BASE_RTP_PAYLOAD_PT(sbcpay));

	payload_data = gst_rtp_buffer_get_payload(outbuf);
	payload = (struct rtp_payload*) payload_data;
	memset(payload, 0, sizeof(struct rtp_payload));
	payload->frame_count = frame_count;

	data = gst_adapter_take(sbcpay->adapter, payload_length);
	memcpy(payload_data + RTP_SBC_PAYLOAD_HEADER_SIZE, data,
			payload_length);
	g_free(data);

	GST_BUFFER_TIMESTAMP(outbuf) = sbcpay->timestamp;
	GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);

	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;

	/* FIXME check for negotiation */

	sbcpay = GST_RTP_SBC_PAY(payload);
	sbcpay->timestamp = GST_BUFFER_TIMESTAMP(buffer);

	gst_adapter_push(sbcpay->adapter, 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->timestamp = 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);
}