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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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If properties are proxied through GBinding this can work only if the proxied property keeps it's own value. The previous implementation will read the original value if the proxied property signals a change and thus nothing will happen.
534 lines
15 KiB
C
534 lines
15 KiB
C
/*
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*
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* BlueZ - Bluetooth protocol stack for Linux
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*
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* Copyright (C) 2012 Collabora Ltd.
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*
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <unistd.h>
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#include <stdint.h>
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#include <string.h>
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#include <poll.h>
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#include <gst/rtp/gstrtppayloads.h>
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#include "gstavdtpsrc.h"
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GST_DEBUG_CATEGORY_STATIC (avdtpsrc_debug);
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#define GST_CAT_DEFAULT (avdtpsrc_debug)
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#define DEFAULT_VOLUME 127
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enum
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{
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PROP_0,
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PROP_TRANSPORT,
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PROP_TRANSPORT_VOLUME,
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};
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#define parent_class gst_avdtp_src_parent_class
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G_DEFINE_TYPE (GstAvdtpSrc, gst_avdtp_src, GST_TYPE_BASE_SRC);
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static GstStaticPadTemplate gst_avdtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\","
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"payload = (int) "
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GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) { 16000, 32000, "
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"44100, 48000 }, " "encoding-name = (string) \"SBC\"; "
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"application/x-rtp, "
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"media = (string) \"audio\","
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"payload = (int) "
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GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) { 8000, 11025, 12000, 16000, "
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"22050, 2400, 32000, 44100, 48000, 64000, 88200, 96000 }, "
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"encoding-name = (string) \"MP4A-LATM\"; "));
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static void gst_avdtp_src_finalize (GObject * object);
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static void gst_avdtp_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_avdtp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static GstCaps *gst_avdtp_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter);
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static gboolean gst_avdtp_src_query (GstBaseSrc * bsrc, GstQuery * query);
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static gboolean gst_avdtp_src_start (GstBaseSrc * bsrc);
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static gboolean gst_avdtp_src_stop (GstBaseSrc * bsrc);
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static GstFlowReturn gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset,
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guint length, GstBuffer ** outbuf);
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static gboolean gst_avdtp_src_unlock (GstBaseSrc * bsrc);
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static gboolean gst_avdtp_src_unlock_stop (GstBaseSrc * bsrc);
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static void
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gst_avdtp_src_class_init (GstAvdtpSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_avdtp_src_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_avdtp_src_set_property);
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gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_avdtp_src_get_property);
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basesrc_class->start = GST_DEBUG_FUNCPTR (gst_avdtp_src_start);
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basesrc_class->stop = GST_DEBUG_FUNCPTR (gst_avdtp_src_stop);
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basesrc_class->create = GST_DEBUG_FUNCPTR (gst_avdtp_src_create);
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basesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_avdtp_src_unlock);
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basesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_avdtp_src_unlock_stop);
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basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_avdtp_src_getcaps);
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basesrc_class->query = GST_DEBUG_FUNCPTR (gst_avdtp_src_query);
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g_object_class_install_property (gobject_class, PROP_TRANSPORT,
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g_param_spec_string ("transport",
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"Transport", "Use configured transport", NULL, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, PROP_TRANSPORT_VOLUME,
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g_param_spec_uint ("transport-volume",
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"Transport volume",
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"Volume of the transport (only valid if transport is acquired)",
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0, 127, DEFAULT_VOLUME, G_PARAM_READWRITE));
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gst_element_class_set_static_metadata (element_class,
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"Bluetooth AVDTP Source",
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"Source/Audio/Network/RTP",
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"Receives audio from an A2DP device",
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"Arun Raghavan <arun.raghavan@collabora.co.uk>");
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GST_DEBUG_CATEGORY_INIT (avdtpsrc_debug, "avdtpsrc", 0,
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"Bluetooth AVDTP Source");
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gst_element_class_add_static_pad_template (element_class,
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&gst_avdtp_src_template);
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}
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static void
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gst_avdtp_src_init (GstAvdtpSrc * avdtpsrc)
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{
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avdtpsrc->poll = gst_poll_new (TRUE);
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avdtpsrc->duration = GST_CLOCK_TIME_NONE;
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avdtpsrc->transport_volume = DEFAULT_VOLUME;
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gst_base_src_set_format (GST_BASE_SRC (avdtpsrc), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (avdtpsrc), TRUE);
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gst_base_src_set_do_timestamp (GST_BASE_SRC (avdtpsrc), TRUE);
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}
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static void
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gst_avdtp_src_finalize (GObject * object)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object);
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gst_poll_free (avdtpsrc->poll);
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gst_avdtp_connection_reset (&avdtpsrc->conn);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_avdtp_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object);
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switch (prop_id) {
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case PROP_TRANSPORT:
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g_value_set_string (value, avdtpsrc->conn.transport);
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break;
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case PROP_TRANSPORT_VOLUME:
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g_value_set_uint (value, avdtpsrc->transport_volume);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_avdtp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (object);
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switch (prop_id) {
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case PROP_TRANSPORT:
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gst_avdtp_connection_set_transport (&avdtpsrc->conn,
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g_value_get_string (value));
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break;
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case PROP_TRANSPORT_VOLUME:
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avdtpsrc->transport_volume = g_value_get_uint (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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gst_avdtp_src_query (GstBaseSrc * bsrc, GstQuery * query)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
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gboolean ret = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_DURATION:{
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GstFormat format;
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if (avdtpsrc->duration != GST_CLOCK_TIME_NONE) {
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gst_query_parse_duration (query, &format, NULL);
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if (format == GST_FORMAT_TIME) {
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gst_query_set_duration (query, format, (gint64) avdtpsrc->duration);
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ret = TRUE;
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}
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}
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break;
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}
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default:
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ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
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}
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return ret;
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}
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static GstCaps *
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gst_avdtp_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
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GstCaps *caps = NULL, *ret = NULL;
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if (avdtpsrc->dev_caps) {
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const GValue *value;
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const char *format;
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int rate;
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GstStructure *structure = gst_caps_get_structure (avdtpsrc->dev_caps, 0);
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format = gst_structure_get_name (structure);
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if (g_str_equal (format, "audio/x-sbc")) {
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/* FIXME: we can return a fixed payload type once we
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* are in PLAYING */
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caps = gst_caps_new_simple ("application/x-rtp",
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"media", G_TYPE_STRING, "audio",
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"payload", GST_TYPE_INT_RANGE, 96, 127,
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"encoding-name", G_TYPE_STRING, "SBC", NULL);
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} else if (g_str_equal (format, "audio/mpeg")) {
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caps = gst_caps_new_simple ("application/x-rtp",
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"media", G_TYPE_STRING, "audio",
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"payload", GST_TYPE_INT_RANGE, 96, 127,
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"encoding-name", G_TYPE_STRING, "MP4A-LATM", NULL);
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value = gst_structure_get_value (structure, "mpegversion");
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if (!value || !G_VALUE_HOLDS_INT (value)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get mpegversion");
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gst_caps_unref (caps);
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return NULL;
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}
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gst_caps_set_simple (caps, "mpegversion", G_TYPE_INT,
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g_value_get_int (value), NULL);
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value = gst_structure_get_value (structure, "channels");
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if (!value || !G_VALUE_HOLDS_INT (value)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get channels");
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gst_caps_unref (caps);
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return NULL;
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}
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gst_caps_set_simple (caps, "channels", G_TYPE_INT,
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g_value_get_int (value), NULL);
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value = gst_structure_get_value (structure, "base-profile");
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if (!value || !G_VALUE_HOLDS_STRING (value)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get base-profile");
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gst_caps_unref (caps);
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return NULL;
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}
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gst_caps_set_simple (caps, "base-profile", G_TYPE_STRING,
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g_value_get_string (value), NULL);
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} else {
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GST_ERROR_OBJECT (avdtpsrc,
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"Only SBC and MPEG-2/4 are supported at the moment");
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}
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value = gst_structure_get_value (structure, "rate");
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if (!value || !G_VALUE_HOLDS_INT (value)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get sample rate");
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gst_caps_unref (caps);
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return NULL;
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}
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rate = g_value_get_int (value);
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gst_caps_set_simple (caps, "clock-rate", G_TYPE_INT, rate, NULL);
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if (filter) {
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ret = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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} else
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ret = caps;
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} else {
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GST_DEBUG_OBJECT (avdtpsrc, "device not open, using template caps");
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ret = GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
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}
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return ret;
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}
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static void
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avrcp_metadata_cb (GstAvrcpConnection * avrcp, GstTagList * taglist,
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gpointer user_data)
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{
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GstAvdtpSrc *src = GST_AVDTP_SRC (user_data);
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guint64 duration;
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if (gst_tag_list_get_uint64 (taglist, GST_TAG_DURATION, &duration)) {
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src->duration = duration;
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gst_element_post_message (GST_ELEMENT (src),
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gst_message_new_duration_changed (GST_OBJECT (src)));
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}
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gst_pad_push_event (GST_BASE_SRC_PAD (src),
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gst_event_new_tag (gst_tag_list_copy (taglist)));
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gst_element_post_message (GST_ELEMENT (src),
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gst_message_new_tag (GST_OBJECT (src), taglist));
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}
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static void
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gst_avdtp_src_start_avrcp (GstAvdtpSrc * src)
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{
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gchar *path, **strv;
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int i;
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/* Strip out the /fdX in /org/bluez/dev_.../fdX */
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strv = g_strsplit (src->conn.transport, "/", -1);
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for (i = 0; strv[i]; i++);
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g_return_if_fail (i > 0);
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g_free (strv[i - 1]);
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strv[i - 1] = NULL;
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path = g_strjoinv ("/", strv);
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g_strfreev (strv);
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src->avrcp = gst_avrcp_connection_new (path, avrcp_metadata_cb, src, NULL);
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g_free (path);
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}
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static void
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gst_avdtp_src_stop_avrcp (GstAvdtpSrc * src)
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{
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gst_avrcp_connection_free (src->avrcp);
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}
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static gboolean
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gst_avdtp_src_start (GstBaseSrc * bsrc)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
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/* None of this can go into prepare() since we need to set up the
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* connection to figure out what format the device is going to send us.
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*/
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if (!gst_avdtp_connection_acquire (&avdtpsrc->conn, FALSE)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to acquire connection");
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return FALSE;
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}
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if (!gst_avdtp_connection_get_properties (&avdtpsrc->conn)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get transport properties");
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goto fail;
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}
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if (!gst_avdtp_connection_conf_recv_stream_fd (&avdtpsrc->conn)) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to configure stream fd");
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goto fail;
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}
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GST_DEBUG_OBJECT (avdtpsrc, "Setting block size to link MTU (%d)",
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avdtpsrc->conn.data.link_mtu);
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gst_base_src_set_blocksize (GST_BASE_SRC (avdtpsrc),
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avdtpsrc->conn.data.link_mtu);
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avdtpsrc->dev_caps = gst_avdtp_connection_get_caps (&avdtpsrc->conn);
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if (!avdtpsrc->dev_caps) {
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GST_ERROR_OBJECT (avdtpsrc, "Failed to get device caps");
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goto fail;
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}
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gst_poll_fd_init (&avdtpsrc->pfd);
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avdtpsrc->pfd.fd = g_io_channel_unix_get_fd (avdtpsrc->conn.stream);
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gst_poll_add_fd (avdtpsrc->poll, &avdtpsrc->pfd);
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gst_poll_fd_ctl_read (avdtpsrc->poll, &avdtpsrc->pfd, TRUE);
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gst_poll_set_flushing (avdtpsrc->poll, FALSE);
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g_atomic_int_set (&avdtpsrc->unlocked, FALSE);
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/* The life time of the connection is shorter than the src object, so we
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* don't need to worry about memory management */
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gst_avdtp_connection_notify_volume (&avdtpsrc->conn, G_OBJECT (avdtpsrc),
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"transport-volume");
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gst_avdtp_src_start_avrcp (avdtpsrc);
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return TRUE;
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fail:
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gst_avdtp_connection_release (&avdtpsrc->conn);
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return FALSE;
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}
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static gboolean
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gst_avdtp_src_stop (GstBaseSrc * bsrc)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
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gst_poll_remove_fd (avdtpsrc->poll, &avdtpsrc->pfd);
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gst_poll_set_flushing (avdtpsrc->poll, TRUE);
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gst_avdtp_src_stop_avrcp (avdtpsrc);
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gst_avdtp_connection_release (&avdtpsrc->conn);
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if (avdtpsrc->dev_caps) {
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gst_caps_unref (avdtpsrc->dev_caps);
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avdtpsrc->dev_caps = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
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GstBuffer ** outbuf)
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{
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GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
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GstBuffer *buf = NULL;
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GstMapInfo info;
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int ret;
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if (g_atomic_int_get (&avdtpsrc->unlocked))
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return GST_FLOW_FLUSHING;
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/* We don't operate in GST_FORMAT_BYTES, so offset is ignored */
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while ((ret = gst_poll_wait (avdtpsrc->poll, GST_CLOCK_TIME_NONE))) {
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if (g_atomic_int_get (&avdtpsrc->unlocked))
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/* We're unlocked, time to gtfo */
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return GST_FLOW_FLUSHING;
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if (ret < 0)
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/* Something went wrong */
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goto read_error;
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if (ret > 0)
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/* Got some data */
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break;
|
|
}
|
|
|
|
ret = GST_BASE_SRC_CLASS (parent_class)->alloc (bsrc, offset, length, outbuf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto alloc_failed;
|
|
|
|
buf = *outbuf;
|
|
|
|
gst_buffer_map (buf, &info, GST_MAP_WRITE);
|
|
|
|
ret = read (avdtpsrc->pfd.fd, info.data, length);
|
|
|
|
if (ret < 0)
|
|
goto read_error;
|
|
else if (ret == 0) {
|
|
GST_INFO_OBJECT (avdtpsrc, "Got EOF on the transport fd");
|
|
goto eof;
|
|
}
|
|
|
|
if (ret < length)
|
|
gst_buffer_set_size (buf, ret);
|
|
|
|
GST_LOG_OBJECT (avdtpsrc, "Read %d bytes", ret);
|
|
|
|
gst_buffer_unmap (buf, &info);
|
|
*outbuf = buf;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
alloc_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (bsrc, "alloc failed: %s", gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
|
|
read_error:
|
|
GST_ERROR_OBJECT (avdtpsrc, "Error while reading audio data: %s",
|
|
strerror (errno));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
|
|
eof:
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
|
|
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);
|
|
}
|