/* * * BlueZ - Bluetooth protocol stack for Linux * * Copyright (C) 2012 Collabora Ltd. * * * 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 #endif #include #include #include #include #include #include "gstavdtpsrc.h" GST_DEBUG_CATEGORY_STATIC (avdtpsrc_debug); #define GST_CAT_DEFAULT (avdtpsrc_debug) enum { PROP_0, PROP_TRANSPORT }; GST_BOILERPLATE (GstAvdtpSrc, gst_avdtp_src, GstBaseSrc, GST_TYPE_BASE_SRC); static GstStaticPadTemplate gst_avdtp_src_template = 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_avdtp_src_finalize (GObject * object); static void gst_avdtp_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_avdtp_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static GstCaps *gst_avdtp_src_getcaps (GstPad * pad); static gboolean gst_avdtp_src_start (GstBaseSrc * bsrc); static gboolean gst_avdtp_src_stop (GstBaseSrc * bsrc); static GstFlowReturn gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, GstBuffer ** outbuf); static gboolean gst_avdtp_src_unlock (GstBaseSrc * bsrc); static gboolean gst_avdtp_src_unlock_stop (GstBaseSrc * bsrc); static void gst_avdtp_src_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_avdtp_src_template)); gst_element_class_set_details_simple (element_class, "Bluetooth AVDTP Source", "Source/Audio/Network/RTP", "Receives audio from an A2DP device", "Arun Raghavan "); } static void gst_avdtp_src_class_init (GstAvdtpSrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass); gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_avdtp_src_finalize); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_avdtp_src_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_avdtp_src_get_property); basesrc_class->start = GST_DEBUG_FUNCPTR (gst_avdtp_src_start); basesrc_class->stop = GST_DEBUG_FUNCPTR (gst_avdtp_src_stop); basesrc_class->create = GST_DEBUG_FUNCPTR (gst_avdtp_src_create); basesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_avdtp_src_unlock); basesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_avdtp_src_unlock_stop); g_object_class_install_property (gobject_class, PROP_TRANSPORT, g_param_spec_string ("transport", "Transport", "Use configured transport", NULL, G_PARAM_READWRITE)); GST_DEBUG_CATEGORY_INIT (avdtpsrc_debug, "avdtpsrc", 0, "Bluetooth AVDTP Source"); } static void gst_avdtp_src_init (GstAvdtpSrc * avdtpsrc, GstAvdtpSrcClass * klass) { avdtpsrc->poll = gst_poll_new (TRUE); gst_base_src_set_format (GST_BASE_SRC (avdtpsrc), GST_FORMAT_DEFAULT); gst_base_src_set_live (GST_BASE_SRC (avdtpsrc), TRUE); gst_base_src_set_do_timestamp (GST_BASE_SRC (avdtpsrc), TRUE); gst_pad_set_getcaps_function (GST_BASE_SRC_PAD (avdtpsrc), GST_DEBUG_FUNCPTR (gst_avdtp_src_getcaps)); } static void gst_avdtp_src_finalize (GObject * object) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object); gst_poll_free (avdtpsrc->poll); gst_avdtp_connection_reset (&avdtpsrc->conn); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_avdtp_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object); switch (prop_id) { case PROP_TRANSPORT: g_value_set_string (value, avdtpsrc->conn.transport); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_avdtp_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object); switch (prop_id) { case PROP_TRANSPORT: gst_avdtp_connection_set_transport (&avdtpsrc->conn, g_value_get_string (value)); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstCaps * gst_avdtp_src_getcaps (GstPad * pad) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (gst_pad_get_parent_element (pad)); GstCaps *ret = NULL; if (avdtpsrc->dev_caps) { const GValue *value; const char *format; int rate; GstStructure *structure = gst_caps_get_structure (avdtpsrc->dev_caps, 0); format = gst_structure_get_name (structure); if (g_str_equal (format, "audio/x-sbc")) { /* FIXME: we can return a fixed payload type once we * are in PLAYING */ ret = gst_caps_new_simple ("application/x-rtp", "media", G_TYPE_STRING, "audio", "payload", GST_TYPE_INT_RANGE, 96, 127, "encoding-name", G_TYPE_STRING, "SBC", NULL); } else if (g_str_equal (format, "audio/mpeg")) { GST_ERROR_OBJECT (avdtpsrc, "Only SBC is supported at " "the moment"); } value = gst_structure_get_value (structure, "rate"); if (!value || !G_VALUE_HOLDS_INT (value)) { GST_ERROR_OBJECT (avdtpsrc, "Failed to get sample rate"); goto fail; } rate = g_value_get_int (value); gst_caps_set_simple (ret, "clock-rate", G_TYPE_INT, rate, NULL); } else { ret = gst_caps_ref (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad))); } return ret; fail: if (ret) gst_caps_unref (ret); return NULL; } static gboolean gst_avdtp_src_start (GstBaseSrc * bsrc) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc); /* None of this can go into prepare() since we need to set up the * connection to figure out what format the device is going to send us. */ if (!gst_avdtp_connection_acquire (&avdtpsrc->conn)) { GST_ERROR_OBJECT (avdtpsrc, "Failed to acquire connection"); return FALSE; } if (!gst_avdtp_connection_get_properties (&avdtpsrc->conn)) { GST_ERROR_OBJECT (avdtpsrc, "Failed to get transport properties"); goto fail; } if (!gst_avdtp_connection_conf_recv_stream_fd (&avdtpsrc->conn)) { GST_ERROR_OBJECT (avdtpsrc, "Failed to configure stream fd"); goto fail; } GST_DEBUG_OBJECT (avdtpsrc, "Setting block size to link MTU (%d)", avdtpsrc->conn.data.link_mtu); gst_base_src_set_blocksize (GST_BASE_SRC (avdtpsrc), avdtpsrc->conn.data.link_mtu); avdtpsrc->dev_caps = gst_avdtp_connection_get_caps (&avdtpsrc->conn); if (!avdtpsrc->dev_caps) { GST_ERROR_OBJECT (avdtpsrc, "Failed to get device caps"); goto fail; } gst_poll_fd_init (&avdtpsrc->pfd); avdtpsrc->pfd.fd = g_io_channel_unix_get_fd (avdtpsrc->conn.stream); gst_poll_add_fd (avdtpsrc->poll, &avdtpsrc->pfd); gst_poll_fd_ctl_read (avdtpsrc->poll, &avdtpsrc->pfd, TRUE); gst_poll_set_flushing (avdtpsrc->poll, FALSE); g_atomic_int_set (&avdtpsrc->unlocked, FALSE); return TRUE; fail: gst_avdtp_connection_release (&avdtpsrc->conn); return FALSE; } static gboolean gst_avdtp_src_stop (GstBaseSrc * bsrc) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc); gst_poll_remove_fd (avdtpsrc->poll, &avdtpsrc->pfd); gst_poll_set_flushing (avdtpsrc->poll, TRUE); gst_avdtp_connection_release (&avdtpsrc->conn); if (avdtpsrc->dev_caps) { gst_caps_unref (avdtpsrc->dev_caps); avdtpsrc->dev_caps = NULL; } return TRUE; } static GstFlowReturn gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset, guint length, GstBuffer ** outbuf) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc); GstBuffer *buf = NULL; int ret; if (g_atomic_int_get (&avdtpsrc->unlocked)) return GST_FLOW_WRONG_STATE; /* We don't operate in GST_FORMAT_BYTES, so offset is ignored */ while ((ret = gst_poll_wait (avdtpsrc->poll, GST_CLOCK_TIME_NONE))) { if (g_atomic_int_get (&avdtpsrc->unlocked)) /* We're unlocked, time to gtfo */ return GST_FLOW_WRONG_STATE; if (ret < 0) /* Something went wrong */ goto read_error; if (ret > 0) /* Got some data */ break; } buf = gst_buffer_new_and_alloc (length); ret = read (avdtpsrc->pfd.fd, GST_BUFFER_DATA (buf), length); if (ret < 0) goto read_error; else if (ret == 0) { GST_ERROR_OBJECT (avdtpsrc, "Got EOF on the transport fd"); GST_ELEMENT_ERROR ((avdtpsrc), RESOURCE, FAILED, ("Disconnected" " from remote device"), (NULL)); goto eof; } GST_LOG_OBJECT (avdtpsrc, "Read %d bytes", ret); if (ret < (gint) length) { /* Create a subbuffer for as much as we've actually read */ *outbuf = gst_buffer_create_sub (buf, 0, ret); gst_buffer_unref (buf); } else *outbuf = buf; return GST_FLOW_OK; read_error: GST_ERROR_OBJECT (avdtpsrc, "Error while reading audio data: %s", strerror (errno)); eof: gst_buffer_unref (buf); return GST_FLOW_ERROR; } static gboolean gst_avdtp_src_unlock (GstBaseSrc * bsrc) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc); g_atomic_int_set (&avdtpsrc->unlocked, TRUE); gst_poll_set_flushing (avdtpsrc->poll, TRUE); return TRUE; } static gboolean gst_avdtp_src_unlock_stop (GstBaseSrc * bsrc) { GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc); g_atomic_int_set (&avdtpsrc->unlocked, FALSE); gst_poll_set_flushing (avdtpsrc->poll, FALSE); /* Flush out any stale data that might be buffered */ gst_avdtp_connection_conf_recv_stream_fd (&avdtpsrc->conn); return TRUE; } gboolean gst_avdtp_src_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "avdtpsrc", GST_RANK_NONE, GST_TYPE_AVDTP_SRC); }