gstreamer/sys/bluez/gstavdtpsink.c
Arun Raghavan f644b924d4 bluez: Add an avdtpsrc element
Source element that connects to a given transport and reads audio over
AVDTP. Does not provide a clock but uses the system clock to timestamp
incoming packets. Only SBC is currently supported.
2013-03-28 16:50:25 +00:00

518 lines
13 KiB
C

/*
*
* BlueZ - Bluetooth protocol stack for Linux
*
* Copyright (C) 2004-2010 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
/* FIXME: check which includes are really required */
#include <unistd.h>
#include <sys/un.h>
#include <sys/socket.h>
#include <fcntl.h>
#include <netinet/in.h>
#include <dbus/dbus.h>
#include "a2dp-codecs.h"
#include "gstavdtpsink.h"
#include <gst/rtp/rtp.h>
GST_DEBUG_CATEGORY_STATIC (avdtp_sink_debug);
#define GST_CAT_DEFAULT avdtp_sink_debug
#define CRC_PROTECTED 1
#define CRC_UNPROTECTED 0
#define DEFAULT_AUTOCONNECT TRUE
#define GST_AVDTP_SINK_MUTEX_LOCK(s) G_STMT_START { \
g_mutex_lock(&s->sink_lock); \
} G_STMT_END
#define GST_AVDTP_SINK_MUTEX_UNLOCK(s) G_STMT_START { \
g_mutex_unlock(&s->sink_lock); \
} G_STMT_END
#define IS_SBC(n) (strcmp((n), "audio/x-sbc") == 0)
#define IS_MPEG_AUDIO(n) (strcmp((n), "audio/mpeg") == 0)
enum
{
PROP_0,
PROP_DEVICE,
PROP_AUTOCONNECT,
PROP_TRANSPORT
};
#define parent_class gst_avdtp_sink_parent_class
G_DEFINE_TYPE (GstAvdtpSink, gst_avdtp_sink, GST_TYPE_BASE_SINK);
static GstStaticPadTemplate avdtp_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, 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\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) "
GST_RTP_PAYLOAD_MPA_STRING ", "
"clock-rate = (int) 90000; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) "
GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\""));
static gboolean
gst_avdtp_sink_stop (GstBaseSink * basesink)
{
GstAvdtpSink *self = GST_AVDTP_SINK (basesink);
GST_INFO_OBJECT (self, "stop");
if (self->watch_id != 0) {
g_source_remove (self->watch_id);
self->watch_id = 0;
}
gst_avdtp_connection_release (&self->conn);
if (self->stream_caps) {
gst_caps_unref (self->stream_caps);
self->stream_caps = NULL;
}
if (self->dev_caps) {
gst_caps_unref (self->dev_caps);
self->dev_caps = NULL;
}
return TRUE;
}
static void
gst_avdtp_sink_finalize (GObject * object)
{
GstAvdtpSink *self = GST_AVDTP_SINK (object);
gst_avdtp_sink_stop (GST_BASE_SINK (self));
gst_avdtp_connection_reset (&self->conn);
g_mutex_clear (&self->sink_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_avdtp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAvdtpSink *sink = GST_AVDTP_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
if (sink->device)
g_free (sink->device);
gst_avdtp_connection_set_device (&sink->conn, g_value_get_string (value));
break;
case PROP_AUTOCONNECT:
sink->autoconnect = g_value_get_boolean (value);
break;
case PROP_TRANSPORT:
if (sink->transport)
g_free (sink->transport);
gst_avdtp_connection_set_transport (&sink->conn,
g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_avdtp_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAvdtpSink *sink = GST_AVDTP_SINK (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, sink->conn.device);
break;
case PROP_AUTOCONNECT:
g_value_set_boolean (value, sink->autoconnect);
break;
case PROP_TRANSPORT:
g_value_set_string (value, sink->conn.transport);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gint
gst_avdtp_sink_get_channel_mode (const gchar * mode)
{
if (strcmp (mode, "stereo") == 0)
return SBC_CHANNEL_MODE_STEREO;
else if (strcmp (mode, "joint-stereo") == 0)
return SBC_CHANNEL_MODE_JOINT_STEREO;
else if (strcmp (mode, "dual-channel") == 0)
return SBC_CHANNEL_MODE_DUAL_CHANNEL;
else if (strcmp (mode, "mono") == 0)
return SBC_CHANNEL_MODE_MONO;
else
return -1;
}
static void
gst_avdtp_sink_tag (const GstTagList * taglist,
const gchar * tag, gpointer user_data)
{
gboolean crc;
gchar *channel_mode = NULL;
GstAvdtpSink *self = GST_AVDTP_SINK (user_data);
if (strcmp (tag, "has-crc") == 0) {
if (!gst_tag_list_get_boolean (taglist, tag, &crc)) {
GST_WARNING_OBJECT (self, "failed to get crc tag");
return;
}
gst_avdtp_sink_set_crc (self, crc);
} else if (strcmp (tag, "channel-mode") == 0) {
if (!gst_tag_list_get_string (taglist, tag, &channel_mode)) {
GST_WARNING_OBJECT (self, "failed to get channel-mode tag");
return;
}
self->channel_mode = gst_avdtp_sink_get_channel_mode (channel_mode);
if (self->channel_mode == -1)
GST_WARNING_OBJECT (self, "Received invalid channel "
"mode: %s", channel_mode);
g_free (channel_mode);
} else
GST_DEBUG_OBJECT (self, "received unused tag: %s", tag);
}
static gboolean
gst_avdtp_sink_event (GstBaseSink * basesink, GstEvent * event)
{
GstAvdtpSink *self = GST_AVDTP_SINK (basesink);
GstTagList *taglist = NULL;
if (GST_EVENT_TYPE (event) == GST_EVENT_TAG) {
/* we check the tags, mp3 has tags that are importants and
* are outside caps */
gst_event_parse_tag (event, &taglist);
gst_tag_list_foreach (taglist, gst_avdtp_sink_tag, self);
}
return TRUE;
}
static gboolean
gst_avdtp_sink_start (GstBaseSink * basesink)
{
GstAvdtpSink *self = GST_AVDTP_SINK (basesink);
GST_INFO_OBJECT (self, "start");
self->stream_caps = NULL;
self->mp3_using_crc = -1;
self->channel_mode = -1;
if (self->conn.transport == NULL)
return FALSE;
if (!gst_avdtp_connection_acquire (&self->conn)) {
GST_ERROR_OBJECT (self, "Failed to acquire connection");
return FALSE;
}
if (!gst_avdtp_connection_get_properties (&self->conn)) {
GST_ERROR_OBJECT (self, "Failed to get transport properties");
return FALSE;
}
if (self->dev_caps)
gst_caps_unref (self->dev_caps);
self->dev_caps = gst_avdtp_connection_get_caps (&self->conn);
if (!self->dev_caps) {
GST_ERROR_OBJECT (self, "Failed to get device caps");
return FALSE;
}
GST_DEBUG_OBJECT (self, "Got connection caps: " GST_PTR_FORMAT,
self->dev_caps);
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_connection_conf_recv_stream_fd (&sink->conn);
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)
{
GstFlowReturn flow_ret = GST_FLOW_OK;
GstAvdtpSink *self = GST_AVDTP_SINK (basesink);
GstMapInfo map;
ssize_t ret;
int fd;
if (!gst_buffer_map (buffer, &map, GST_MAP_READ))
return GST_FLOW_ERROR;
/* FIXME: temporary sanity check */
g_assert (!(g_io_channel_get_flags (self->stream) & G_IO_FLAG_NONBLOCK));
/* FIXME: why not use g_io_channel_write_chars() instead? */
fd = g_io_channel_unix_get_fd (self->conn.stream);
ret = write (fd, map.data, map.size);
if (ret < 0) {
/* FIXME: since this is probably fatal, shouldn't we post an error here? */
GST_ERROR_OBJECT (self, "Error writing to socket: %s", g_strerror (errno));
flow_ret = GST_FLOW_ERROR;
}
gst_buffer_unmap (buffer, &map);
return flow_ret;
}
static gboolean
gst_avdtp_sink_unlock (GstBaseSink * basesink)
{
GstAvdtpSink *self = GST_AVDTP_SINK (basesink);
if (self->conn.stream != NULL)
g_io_channel_flush (self->conn.stream, NULL);
return TRUE;
}
static void
gst_avdtp_sink_class_init (GstAvdtpSinkClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_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);
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));
g_object_class_install_property (object_class, PROP_TRANSPORT,
g_param_spec_string ("transport",
"Transport", "Use configured transport", NULL, G_PARAM_READWRITE));
GST_DEBUG_CATEGORY_INIT (avdtp_sink_debug, "avdtpsink", 0,
"A2DP headset sink element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&avdtp_sink_factory));
gst_element_class_set_static_metadata (element_class, "Bluetooth AVDTP sink",
"Sink/Audio", "Plays audio to an A2DP device",
"Marcel Holtmann <marcel@holtmann.org>");
}
static void
gst_avdtp_sink_init (GstAvdtpSink * self)
{
self->conn.device = NULL;
self->conn.transport = NULL;
self->conn.stream = NULL;
self->dev_caps = NULL;
self->autoconnect = DEFAULT_AUTOCONNECT;
g_mutex_init (&self->sink_lock);
/* 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);
*/
}
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)
{
GST_DEBUG_OBJECT (self, "setting device caps");
GST_AVDTP_SINK_MUTEX_LOCK (self);
if (self->stream_caps)
gst_caps_unref (self->stream_caps);
self->stream_caps = gst_caps_ref (caps);
GST_AVDTP_SINK_MUTEX_UNLOCK (self);
return TRUE;
}
guint
gst_avdtp_sink_get_link_mtu (GstAvdtpSink * sink)
{
return sink->conn.data.link_mtu;
}
void
gst_avdtp_sink_set_device (GstAvdtpSink * self, const gchar * dev)
{
if (self->conn.device != NULL)
g_free (self->conn.device);
GST_LOG_OBJECT (self, "Setting device: %s", dev);
self->conn.device = g_strdup (dev);
}
void
gst_avdtp_sink_set_transport (GstAvdtpSink * self, const gchar * trans)
{
if (self->conn.transport != NULL)
g_free (self->conn.transport);
GST_LOG_OBJECT (self, "Setting transport: %s", trans);
self->conn.transport = g_strdup (trans);
}
gchar *
gst_avdtp_sink_get_device (GstAvdtpSink * self)
{
return g_strdup (self->conn.device);
}
gchar *
gst_avdtp_sink_get_transport (GstAvdtpSink * self)
{
return g_strdup (self->conn.transport);
}
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);
}