gstreamer/sys/bluez/gstavdtpsrc.c
Bernhard Miller 597e3cc98d bluez: support aac in avdtpsrc
Signed-off-by: Bernhard Miller <bernhard.miller@streamunlimited.com>
2013-08-29 10:17:07 +02:00

429 lines
12 KiB
C

/*
*
* 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 <config.h>
#endif
#include <unistd.h>
#include <stdint.h>
#include <string.h>
#include <poll.h>
#include <gst/rtp/gstrtppayloads.h>
#include "gstavdtpsrc.h"
GST_DEBUG_CATEGORY_STATIC (avdtpsrc_debug);
#define GST_CAT_DEFAULT (avdtpsrc_debug)
enum
{
PROP_0,
PROP_TRANSPORT
};
#define parent_class gst_avdtp_src_parent_class
G_DEFINE_TYPE (GstAvdtpSrc, gst_avdtp_src, 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\"; "
"application/x-rtp, "
"media = (string) \"audio\","
"payload = (int) "
GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 8000, 11025, 12000, 16000, "
"22050, 2400, 32000, 44100, 48000, 64000, 88200, 96000 }, "
"encoding-name = (string) \"MP4A-LATM\"; "));
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 (GstBaseSrc * bsrc, GstCaps * filter);
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_class_init (GstAvdtpSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
parent_class = g_type_class_peek_parent (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);
basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_avdtp_src_getcaps);
g_object_class_install_property (gobject_class, PROP_TRANSPORT,
g_param_spec_string ("transport",
"Transport", "Use configured transport", NULL, G_PARAM_READWRITE));
gst_element_class_set_static_metadata (element_class,
"Bluetooth AVDTP Source",
"Source/Audio/Network/RTP",
"Receives audio from an A2DP device",
"Arun Raghavan <arun.raghavan@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (avdtpsrc_debug, "avdtpsrc", 0,
"Bluetooth AVDTP Source");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_avdtp_src_template));
}
static void
gst_avdtp_src_init (GstAvdtpSrc * avdtpsrc)
{
avdtpsrc->poll = gst_poll_new (TRUE);
gst_base_src_set_format (GST_BASE_SRC (avdtpsrc), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (avdtpsrc), TRUE);
gst_base_src_set_do_timestamp (GST_BASE_SRC (avdtpsrc), TRUE);
}
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 (GstBaseSrc * bsrc, GstCaps * filter)
{
GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
GstCaps *caps = NULL, *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 */
caps = 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")) {
caps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"payload", GST_TYPE_INT_RANGE, 96, 127,
"encoding-name", G_TYPE_STRING, "MP4A-LATM", NULL);
value = gst_structure_get_value (structure, "mpegversion");
if (!value || !G_VALUE_HOLDS_INT (value)) {
GST_ERROR_OBJECT (avdtpsrc, "Failed to get mpegversion");
goto fail;
}
gst_caps_set_simple (caps, "mpegversion", G_TYPE_INT,
g_value_get_int (value), NULL);
value = gst_structure_get_value (structure, "channels");
if (!value || !G_VALUE_HOLDS_INT (value)) {
GST_ERROR_OBJECT (avdtpsrc, "Failed to get channels");
goto fail;
}
gst_caps_set_simple (caps, "channels", G_TYPE_INT,
g_value_get_int (value), NULL);
value = gst_structure_get_value (structure, "base-profile");
if (!value || !G_VALUE_HOLDS_STRING (value)) {
GST_ERROR_OBJECT (avdtpsrc, "Failed to get base-profile");
goto fail;
}
gst_caps_set_simple (caps, "base-profile", G_TYPE_STRING,
g_value_get_string (value), NULL);
} else {
GST_ERROR_OBJECT (avdtpsrc,
"Only SBC and MPEG-2/4 are 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 (caps, "clock-rate", G_TYPE_INT, rate, NULL);
if (filter) {
ret = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
} else
ret = caps;
} else {
GST_DEBUG_OBJECT (avdtpsrc, "device not open, using template caps");
ret = GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
}
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;
GstMapInfo info;
int ret;
if (g_atomic_int_get (&avdtpsrc->unlocked))
return GST_FLOW_FLUSHING;
/* 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_FLUSHING;
if (ret < 0)
/* Something went wrong */
goto read_error;
if (ret > 0)
/* Got some data */
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);
}