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audiopayload: add support for buffer-lists

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This commit is contained in:
Wim Taymans 2010-01-06 13:39:14 +01:00
parent 951b8516f1
commit 73d5ae1107
1 changed files with 143 additions and 12 deletions
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155
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -70,6 +70,15 @@
GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
#define DEFAULT_BUFFER_LIST FALSE
enum
{
PROP_0,
PROP_BUFFER_LIST,
PROP_LAST
};
/* function to convert bytes to a time */
typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload,
guint64 bytes);
@ -101,6 +110,8 @@ struct _GstBaseRTPAudioPayloadPrivate
guint cached_ptime;
guint cached_min_length;
guint cached_max_length;
gboolean buffer_list;
};
@ -110,6 +121,11 @@ struct _GstBaseRTPAudioPayloadPrivate
static void gst_base_rtp_audio_payload_finalize (GObject * object);
static void gst_base_rtp_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_base_rtp_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* bytes to time functions */
static GstClockTime
gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
@ -165,6 +181,13 @@ gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize = gst_base_rtp_audio_payload_finalize;
gobject_class->set_property = gst_base_rtp_audio_payload_set_property;
gobject_class->get_property = gst_base_rtp_audio_payload_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
g_param_spec_boolean ("buffer-list", "Buffer List",
"Use Buffer Lists",
DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state);
@ -192,6 +215,8 @@ gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * payload,
payload->sample_size = 0;
payload->priv->adapter = gst_adapter_new ();
payload->priv->buffer_list = DEFAULT_BUFFER_LIST;
}
static void
@ -206,6 +231,42 @@ gst_base_rtp_audio_payload_finalize (GObject * object)
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
static void
gst_base_rtp_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstBaseRTPAudioPayload *payload;
payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
switch (prop_id) {
case PROP_BUFFER_LIST:
payload->priv->buffer_list = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_base_rtp_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstBaseRTPAudioPayload *payload;
payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
switch (prop_id) {
case PROP_BUFFER_LIST:
g_value_set_boolean (value, payload->priv->buffer_list);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/**
* gst_base_rtp_audio_payload_set_frame_based:
* @basertpaudiopayload: a pointer to the element.
@ -414,6 +475,68 @@ gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
return ret;
}
static GstFlowReturn
gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
baseaudiopayload, GstBuffer * buffer)
{
GstBaseRTPPayload *basepayload;
GstBaseRTPAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstClockTime timestamp;
guint8 *payload;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
payload_len = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) {
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
}
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
GstBufferListIterator *it;
list = gst_buffer_list_new ();
it = gst_buffer_list_iterate (list);
/* add both buffers to the buffer list */
gst_buffer_list_iterator_add_group (it);
gst_buffer_list_iterator_add (it, outbuf);
gst_buffer_list_iterator_add (it, buffer);
gst_buffer_list_iterator_free (it);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_basertppayload_push_list (basepayload, list);
} else {
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, GST_BUFFER_DATA (buffer), payload_len);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}
/**
* gst_base_rtp_audio_payload_flush:
* @baseaudiopayload: a #GstBaseRTPPayload
@ -473,18 +596,28 @@ gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {
GstBuffer *buffer;
/* we can quickly take a buffer out of the adapter without having to copy
* anything. */
buffer = gst_adapter_take_buffer (adapter, payload_len);
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (adapter, payload, 0, payload_len);
gst_adapter_flush (adapter, payload_len);
ret = gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer);
} else {
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
/* copy payload */
payload = gst_rtp_buffer_get_payload (outbuf);
gst_adapter_copy (adapter, payload, 0, payload_len);
gst_adapter_flush (adapter, payload_len);
ret = gst_basertppayload_push (basepayload, outbuf);
/* set metadata */
gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_basertppayload_push (basepayload, outbuf);
}
return ret;
}
@ -720,9 +853,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
/* If buffer fits on an RTP packet, let's just push it through
* this will check against max_ptime and max_mtu */
GST_DEBUG_OBJECT (payload, "Fast packet push");
ret = gst_base_rtp_audio_payload_push (payload,
GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
ret = gst_base_rtp_audio_payload_push_buffer (payload, buffer);
} else {
/* push the buffer in the adapter */
gst_adapter_push (priv->adapter, buffer);