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bluez: Compile fixes for avdtpsrc/a2dpsrc code against 1.x

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This commit is contained in:
Arun Raghavan 2013-03-14 18:04:55 +05:30
parent adc9d39710
commit a3713f7be1
3 changed files with 72 additions and 59 deletions
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32
sys/bluez/gsta2dpsrc.c
View file

@ -33,7 +33,8 @@ enum
PROP_TRANSPORT
};
GST_BOILERPLATE (GstA2dpSrc, gst_a2dp_src, GstBin, GST_TYPE_BIN);
#define parent_class gst_a2dp_src_parent_class
G_DEFINE_TYPE (GstA2dpSrc, gst_a2dp_src, GST_TYPE_BIN);
static void gst_a2dp_src_finalize (GObject * object);
static void gst_a2dp_src_get_property (GObject * object, guint prop_id,
@ -53,25 +54,13 @@ GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
"allocation = (string) { \"snr\", \"loudness\" }, "
"bitpool = (int) [ 2, " TEMPLATE_MAX_BITPOOL_STR " ]"));
static void
gst_a2dp_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_a2dp_src_template));
gst_element_class_set_details_simple (element_class,
"Bluetooth A2DP Source",
"Source/Audio/Network",
"Receives and depayloads audio from an A2DP device",
"Arun Raghavan <arun.raghavan@collabora.co.uk>");
}
static void
gst_a2dp_src_class_init (GstA2dpSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_a2dp_src_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_a2dp_src_set_property);
@ -81,12 +70,21 @@ gst_a2dp_src_class_init (GstA2dpSrcClass * klass)
g_param_spec_string ("transport",
"Transport", "Use configured transport", NULL, G_PARAM_READWRITE));
gst_element_class_set_static_metadata (element_class,
"Bluetooth A2DP Source",
"Source/Audio/Network",
"Receives and depayloads audio from an A2DP device",
"Arun Raghavan <arun.raghavan@collabora.co.uk>");
GST_DEBUG_CATEGORY_INIT (a2dpsrc_debug, "a2dpsrc", 0,
"Bluetooth A2DP Source");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_a2dp_src_template));
}
static void
gst_a2dp_src_init (GstA2dpSrc * a2dpsrc, GstA2dpSrcClass * klass)
gst_a2dp_src_init (GstA2dpSrc * a2dpsrc)
{
GstBin *bin = GST_BIN (a2dpsrc);
GstElement *depay = NULL;

95
sys/bluez/gstavdtpsrc.c
View file

@ -42,7 +42,8 @@ enum
PROP_TRANSPORT
};
GST_BOILERPLATE (GstAvdtpSrc, gst_avdtp_src, GstBaseSrc, GST_TYPE_BASE_SRC);
#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,
@ -59,7 +60,7 @@ static void gst_avdtp_src_get_property (GObject * object, guint prop_id,
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 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,
@ -67,27 +68,15 @@ static GstFlowReturn gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset,
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 <arun.raghavan@collabora.co.uk>");
}
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);
@ -97,26 +86,33 @@ gst_avdtp_src_class_init (GstAvdtpSrcClass * klass)
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, GstAvdtpSrcClass * klass)
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);
gst_pad_set_getcaps_function (GST_BASE_SRC_PAD (avdtpsrc),
GST_DEBUG_FUNCPTR (gst_avdtp_src_getcaps));
}
static void
@ -167,10 +163,10 @@ gst_avdtp_src_set_property (GObject * object, guint prop_id,
}
static GstCaps *
gst_avdtp_src_getcaps (GstPad * pad)
gst_avdtp_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (gst_pad_get_parent_element (pad));
GstCaps *ret = NULL;
GstAvdtpSrc *avdtpsrc = GST_AVDTP_SRC (bsrc);
GstCaps *caps = NULL, *ret = NULL;
if (avdtpsrc->dev_caps) {
const GValue *value;
@ -183,7 +179,7 @@ gst_avdtp_src_getcaps (GstPad * pad)
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",
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);
@ -198,9 +194,16 @@ gst_avdtp_src_getcaps (GstPad * pad)
}
rate = g_value_get_int (value);
gst_caps_set_simple (ret, "clock-rate", G_TYPE_INT, rate, NULL);
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 {
ret = gst_caps_ref (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
GST_DEBUG_OBJECT (avdtpsrc, "device not open, using template caps");
ret = GST_BASE_SRC_CLASS (parent_class)->get_caps (bsrc, filter);
}
return ret;
@ -282,22 +285,23 @@ gst_avdtp_src_stop (GstBaseSrc * bsrc)
}
static GstFlowReturn
gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset,
guint length, GstBuffer ** outbuf)
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_WRONG_STATE;
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_WRONG_STATE;
return GST_FLOW_FLUSHING;
if (ret < 0)
/* Something went wrong */
@ -308,8 +312,15 @@ gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset,
break;
}
buf = gst_buffer_new_and_alloc (length);
ret = read (avdtpsrc->pfd.fd, GST_BUFFER_DATA (buf), length);
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;
@ -320,21 +331,25 @@ gst_avdtp_src_create (GstBaseSrc * bsrc, guint64 offset,
goto eof;
}
if (ret < length)
gst_buffer_set_size (buf, ret);
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;
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));
eof:
gst_buffer_unref (buf);

4
sys/bluez/gstavdtputil.c
View file

@ -319,7 +319,7 @@ gst_avdtp_util_parse_sbc_raw (void *config)
GValue *list;
gboolean mono, stereo;
structure = gst_structure_empty_new ("audio/x-sbc");
structure = gst_structure_new_empty ("audio/x-sbc");
value = g_value_init (g_new0 (GValue, 1), G_TYPE_STRING);
list = g_value_init (g_new0 (GValue, 1), GST_TYPE_LIST);
@ -490,7 +490,7 @@ gst_avdtp_util_parse_mpeg_raw (void *config)
gboolean valid_layer = FALSE;
gboolean mono, stereo;
structure = gst_structure_empty_new ("audio/mpeg");
structure = gst_structure_new_empty ("audio/mpeg");
value = g_new0 (GValue, 1);
g_value_init (value, G_TYPE_INT);