gstavdtpsink.c

这是Linux环境下的蓝牙源代码

	GST_DEBUG_OBJECT(self, "stream start packet sent");

	io_error = gst_avdtp_sink_audioservice_expect(self,
			&rsp->rsp_h.msg_h, BT_STREAMSTART_RSP);
	if (io_error != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error while stream "
			"start confirmation");
		return FALSE;
	}

	if (rsp->rsp_h.posix_errno != 0) {
		GST_ERROR_OBJECT(self, "BT_STREAMSTART_RSP failed : %s(%d)",
					strerror(rsp->rsp_h.posix_errno),
					rsp->rsp_h.posix_errno);
		return FALSE;
	}

	GST_DEBUG_OBJECT(self, "stream started");

	io_error = gst_avdtp_sink_audioservice_expect(self, &ind->h,
			BT_STREAMFD_IND);
	if (io_error != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error while receiving "
			"stream filedescriptor");
		return FALSE;
	}

	if (!gst_avdtp_sink_conf_recv_stream_fd(self))
		return FALSE;

	return TRUE;
}

static gboolean gst_avdtp_sink_init_mp3_pkt_conf(
		GstAvdtpSink *self, GstCaps *caps,
		mpeg_capabilities_t *pkt)
{
	const GValue *value = NULL;
	gint rate, layer;
	const gchar* name;
	GstStructure *structure = gst_caps_get_structure(caps, 0);

	name = gst_structure_get_name(structure);

	if (!(IS_MPEG_AUDIO(name))) {
		GST_ERROR_OBJECT(self, "Unexpected format %s, "
				"was expecting mp3", name);
		return FALSE;
	}

	/* layer */
	value = gst_structure_get_value(structure, "layer");
	layer = g_value_get_int(value);
	if (layer == 1)
		pkt->layer = BT_MPEG_LAYER_1;
	else if (layer == 2)
		pkt->layer = BT_MPEG_LAYER_2;
	else if (layer == 3)
		pkt->layer = BT_MPEG_LAYER_3;
	else {
		GST_ERROR_OBJECT(self, "Unexpected layer: %d", layer);
		return FALSE;
	}

	/* crc */
	if (self->mp3_using_crc != -1)
		pkt->crc = self->mp3_using_crc;
	else {
		GST_ERROR_OBJECT(self, "No info about crc was received, "
				" can't proceed");
		return FALSE;
	}

	/* channel mode */
	if (self->channel_mode != -1)
		pkt->channel_mode = self->channel_mode;
	else {
		GST_ERROR_OBJECT(self, "No info about channel mode "
				"received, can't proceed");
		return FALSE;
	}

	/* mpf - we will only use the mandatory one */
	pkt->mpf = 0;

	value = gst_structure_get_value(structure, "rate");
	rate = g_value_get_int(value);
	if (rate == 44100)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_44100;
	else if (rate == 48000)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_48000;
	else if (rate == 32000)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_32000;
	else if (rate == 24000)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_24000;
	else if (rate == 22050)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_22050;
	else if (rate == 16000)
		pkt->frequency = BT_MPEG_SAMPLING_FREQ_16000;
	else {
		GST_ERROR_OBJECT(self, "Invalid rate while setting caps");
		return FALSE;
	}

	/* vbr - we always say its vbr, we don't have how to know it */
	pkt->bitrate = 0x8000;

	return TRUE;
}

static gboolean gst_avdtp_sink_configure(GstAvdtpSink *self,
			GstCaps *caps)
{
	gchar buf[BT_AUDIO_IPC_PACKET_SIZE];
	struct bt_setconfiguration_req *req = (void *) buf;
	struct bt_setconfiguration_rsp *rsp = (void *) buf;
	gboolean ret;
	GIOError io_error;
	gchar *temp;
	GstStructure *structure;

	temp = gst_caps_to_string(caps);
	GST_DEBUG_OBJECT(self, "configuring device with caps: %s", temp);
	g_free(temp);

	memset (req, 0, sizeof(buf));
	req->h.msg_type = BT_SETCONFIGURATION_REQ;
	req->access_mode = BT_CAPABILITIES_ACCESS_MODE_WRITE;
	strncpy(req->device, self->device, 18);
	structure = gst_caps_get_structure(caps, 0);

	if (gst_structure_has_name(structure, "audio/x-sbc"))
		ret = gst_avdtp_sink_init_sbc_pkt_conf(self, caps,
				&req->sbc_capabilities);
	else if (gst_structure_has_name(structure, "audio/mpeg"))
		ret = gst_avdtp_sink_init_mp3_pkt_conf(self, caps,
				&req->mpeg_capabilities);
	else
		ret = FALSE;

	if (!ret) {
		GST_ERROR_OBJECT(self, "Couldn't parse caps "
				"to packet configuration");
		return FALSE;
	}

	io_error = gst_avdtp_sink_audioservice_send(self, &req->h);
	if (io_error != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error ocurred while sending "
					"configurarion packet");
		return FALSE;
	}

	GST_DEBUG_OBJECT(self, "configuration packet sent");

	io_error = gst_avdtp_sink_audioservice_expect(self,
			&rsp->rsp_h.msg_h, BT_SETCONFIGURATION_RSP);
	if (io_error != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error while receiving device "
					"confirmation");
		return FALSE;
	}

	if (rsp->rsp_h.posix_errno != 0) {
		GST_ERROR_OBJECT(self, "BT_SETCONFIGURATION_RSP failed : "
					"%s(%d)",
					strerror(rsp->rsp_h.posix_errno),
					rsp->rsp_h.posix_errno);
		return FALSE;
	}

	self->data->link_mtu = rsp->link_mtu;
	GST_DEBUG_OBJECT(self, "configuration set");

	return TRUE;
}

static GstFlowReturn gst_avdtp_sink_preroll(GstBaseSink *basesink,
					GstBuffer *buffer)
{
	GstAvdtpSink *sink = GST_AVDTP_SINK(basesink);
	gboolean ret;

	GST_AVDTP_SINK_MUTEX_LOCK(sink);

	ret = gst_avdtp_sink_stream_start(sink);

	GST_AVDTP_SINK_MUTEX_UNLOCK(sink);

	if (!ret)
		return GST_FLOW_ERROR;

	return GST_FLOW_OK;
}

static GstFlowReturn gst_avdtp_sink_render(GstBaseSink *basesink,
					GstBuffer *buffer)
{
	GstAvdtpSink *self = GST_AVDTP_SINK(basesink);
	gsize ret;
	GIOError err;

	err = g_io_channel_write(self->stream, (gchar*)GST_BUFFER_DATA(buffer),
			 (gsize)(GST_BUFFER_SIZE(buffer)), &ret);

	if (err != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error while writting to socket: %d %s",
				errno, strerror(errno));
		return GST_FLOW_ERROR;
	}

	return GST_FLOW_OK;
}

static gboolean gst_avdtp_sink_unlock(GstBaseSink *basesink)
{
	GstAvdtpSink *self = GST_AVDTP_SINK(basesink);

	if (self->stream != NULL)
		g_io_channel_flush (self->stream, NULL);

	return TRUE;
}

static GstFlowReturn gst_avdtp_sink_buffer_alloc(GstBaseSink *basesink,
				guint64 offset, guint size, GstCaps* caps,
				GstBuffer **buf)
{
	GstAvdtpSink *self = GST_AVDTP_SINK(basesink);

	*buf = gst_buffer_new_and_alloc(size);
	if (!(*buf)) {
		GST_ERROR_OBJECT(self, "buffer allocation failed");
		return GST_FLOW_ERROR;
	}

	gst_buffer_set_caps(*buf, caps);

	GST_BUFFER_OFFSET(*buf) = offset;

	return GST_FLOW_OK;
}

static void gst_avdtp_sink_class_init(GstAvdtpSinkClass *klass)
{
	GObjectClass *object_class = G_OBJECT_CLASS(klass);
	GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS(klass);

	parent_class = g_type_class_peek_parent(klass);

	object_class->finalize = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_finalize);
	object_class->set_property = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_set_property);
	object_class->get_property = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_get_property);

	basesink_class->start = GST_DEBUG_FUNCPTR(gst_avdtp_sink_start);
	basesink_class->stop = GST_DEBUG_FUNCPTR(gst_avdtp_sink_stop);
	basesink_class->render = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_render);
	basesink_class->preroll = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_preroll);
	basesink_class->unlock = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_unlock);
	basesink_class->event = GST_DEBUG_FUNCPTR(
					gst_avdtp_sink_event);

	basesink_class->buffer_alloc =
		GST_DEBUG_FUNCPTR(gst_avdtp_sink_buffer_alloc);

	g_object_class_install_property(object_class, PROP_DEVICE,
					g_param_spec_string("device", "Device",
					"Bluetooth remote device address",
					NULL, G_PARAM_READWRITE));

	g_object_class_install_property(object_class, PROP_AUTOCONNECT,
					g_param_spec_boolean("auto-connect",
					"Auto-connect",
					"Automatically attempt to connect "
					"to device", DEFAULT_AUTOCONNECT,
					G_PARAM_READWRITE));

	GST_DEBUG_CATEGORY_INIT(avdtp_sink_debug, "avdtpsink", 0,
				"A2DP headset sink element");
}

static void gst_avdtp_sink_init(GstAvdtpSink *self,
			GstAvdtpSinkClass *klass)
{
	self->device = NULL;
	self->data = NULL;

	self->stream = NULL;

	self->dev_caps = NULL;

	self->autoconnect = DEFAULT_AUTOCONNECT;

	self->sink_lock = g_mutex_new();

	/* FIXME this is for not synchronizing with clock, should be tested
	 * with devices to see the behaviour
	gst_base_sink_set_sync(GST_BASE_SINK(self), FALSE);
	*/
}

static GIOError gst_avdtp_sink_audioservice_send(
					GstAvdtpSink *self,
					const bt_audio_msg_header_t *msg)
{
	GIOError error;
	gsize written;

	error = g_io_channel_write(self->server, (const gchar*) msg,
			BT_AUDIO_IPC_PACKET_SIZE, &written);
	if (error != G_IO_ERROR_NONE)
		GST_ERROR_OBJECT(self, "Error sending data to audio service:"
			" %s(%d)", strerror(errno), errno);

	return error;
}

static GIOError gst_avdtp_sink_audioservice_recv(
					GstAvdtpSink *self,
					bt_audio_msg_header_t *inmsg)
{
	GIOError status;
	gsize bytes_read;
	const char *type;

	status = g_io_channel_read(self->server, (gchar*) inmsg,
			BT_AUDIO_IPC_PACKET_SIZE, &bytes_read);
	if (status != G_IO_ERROR_NONE) {
		GST_ERROR_OBJECT(self, "Error receiving data from "
				"audio service");
		return status;
	}

	type = bt_audio_strmsg(inmsg->msg_type);
	if (!type) {
		status = G_IO_ERROR_INVAL;
		GST_ERROR_OBJECT(self, "Bogus message type %d "
				"received from audio service",
				inmsg->msg_type);
	}

	return status;
}

static GIOError gst_avdtp_sink_audioservice_expect(
			GstAvdtpSink *self, bt_audio_msg_header_t *outmsg,
			int expected_type)
{
	GIOError status;

	status = gst_avdtp_sink_audioservice_recv(self, outmsg);
	if (status != G_IO_ERROR_NONE)
		return status;

	if (outmsg->msg_type != expected_type)
		status = G_IO_ERROR_INVAL;

	return status;
}

gboolean gst_avdtp_sink_plugin_init (GstPlugin * plugin)
{
	return gst_element_register (plugin, "avdtpsink",
			GST_RANK_NONE, GST_TYPE_AVDTP_SINK);
}


/* public functions */
GstCaps *gst_avdtp_sink_get_device_caps(GstAvdtpSink *sink)
{
	if (sink->dev_caps == NULL)
		return NULL;

	return gst_caps_copy(sink->dev_caps);
}

gboolean gst_avdtp_sink_set_device_caps(GstAvdtpSink *self,
			GstCaps *caps)
{
	gboolean ret;

	GST_DEBUG_OBJECT(self, "setting device caps");
	GST_AVDTP_SINK_MUTEX_LOCK(self);
	ret = gst_avdtp_sink_configure(self, caps);

	if (self->stream_caps)
		gst_caps_unref(self->stream_caps);
	self->stream_caps = gst_caps_ref(caps);

	GST_AVDTP_SINK_MUTEX_UNLOCK(self);

	return ret;
}

guint gst_avdtp_sink_get_link_mtu(GstAvdtpSink *sink)
{
	return sink->data->link_mtu;
}

void gst_avdtp_sink_set_device(GstAvdtpSink *self, const gchar* dev)
{
	if (self->device != NULL)
		g_free(self->device);

	GST_LOG_OBJECT(self, "Setting device: %s", dev);
	self->device = g_strdup(dev);
}

gchar *gst_avdtp_sink_get_device(GstAvdtpSink *self)
{
	return g_strdup(self->device);
}

void gst_avdtp_sink_set_crc(GstAvdtpSink *self, gboolean crc)
{
	gint new_crc;

	new_crc = crc ? CRC_PROTECTED : CRC_UNPROTECTED;

	/* test if we already received a different crc */
	if (self->mp3_using_crc != -1 && new_crc != self->mp3_using_crc) {
		GST_WARNING_OBJECT(self, "crc changed during stream");
		return;
	}
	self->mp3_using_crc = new_crc;

}

void gst_avdtp_sink_set_channel_mode(GstAvdtpSink *self,
			const gchar *mode)
{
	gint new_mode;

	new_mode = gst_avdtp_sink_get_channel_mode(mode);

	if (self->channel_mode != -1 && new_mode != self->channel_mode) {
		GST_WARNING_OBJECT(self, "channel mode changed during stream");
		return;
	}

	self->channel_mode = new_mode;
	if (self->channel_mode == -1)
		GST_WARNING_OBJECT(self, "Received invalid channel "
				"mode: %s", mode);
}