mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 18:51:11 +00:00
516 lines
13 KiB
C
516 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
|
|
|
|
#define GLIB_DISABLE_DEPRECATION_WARNINGS
|
|
|
|
/* 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:
|
|
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:
|
|
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->conn.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);
|
|
}
|