mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 16:50:47 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv

Browse source 
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
This commit is contained in:
Philippe Kalaf 2006-09-27 00:13:29 +00:00
parent f5a74b2643
commit 214a128382
3 changed files with 79 additions and 54 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

7
ChangeLog
View file

@ -1,3 +1,10 @@
2006-09-26 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
2006-09-25 Wim Taymans <wim@fluendo.com>
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),

102
gst-libs/gst/rtp/gstbasertpaudiopayload.c
View file

@ -30,60 +30,80 @@
GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
static void gst_basertpaudiopayload_finalize (GObject * object);
typedef enum
{
AUDIO_CODEC_TYPE_NONE,
AUDIO_CODEC_TYPE_FRAME_BASED,
AUDIO_CODEC_TYPE_SAMPLE_BASED
} AudioCodecType;
struct _GstBaseRTPAudioPayloadPrivate
{
AudioCodecType type;
};
#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \
(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_BASE_RTP_AUDIO_PAYLOAD, \
GstBaseRTPAudioPayloadPrivate))
static void gst_base_rtp_audio_payload_finalize (GObject * object);
static GstFlowReturn
gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload, guint8 * data,
guint payload_len, GstClockTime timestamp);
static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
* payload, GstBuffer * buffer);
static GstFlowReturn
gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
static GstFlowReturn
gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload,
GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_basertpaudiopayload_base_init (gpointer klass)
gst_base_rtp_audio_payload_base_init (gpointer klass)
{
}
static void
gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass)
gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate));
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize =
GST_DEBUG_FUNCPTR (gst_basertpaudiopayload_finalize);
GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_finalize);
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_basertpaudiopayload_handle_buffer);
GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer);
GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
"base audio RTP payloader");
}
static void
gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
GstBaseRTPAudioPayloadClass * klass)
{
basertpaudiopayload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (self);
basertpaudiopayload->base_ts = 0;
basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_NONE;
/* these need to be set by child object if frame based */
basertpaudiopayload->frame_size = 0;
@ -94,7 +114,7 @@ gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
}
static void
gst_basertpaudiopayload_finalize (GObject * object)
gst_base_rtp_audio_payload_finalize (GObject * object)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
@ -104,7 +124,7 @@ gst_basertpaudiopayload_finalize (GObject * object)
}
/**
* gst_basertpaudiopayload_set_frame_based:
* gst_base_rtp_audio_payload_set_frame_based:
* @basertpaudiopayload: a pointer to the element.
*
* Tells #GstBaseRTPAudioPayload that the child element is for a frame based
@ -112,18 +132,18 @@ gst_basertpaudiopayload_finalize (GObject * object)
*
*/
void
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *
gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->type == AUDIO_CODEC_TYPE_NONE);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED;
}
/**
* gst_basertpaudiopayload_set_sample_based:
* gst_base_rtp_audio_payload_set_sample_based:
* @basertpaudiopayload: a pointer to the element.
*
* Tells #GstBaseRTPAudioPayload that the child element is for a sample based
@ -131,18 +151,18 @@ gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *
*
*/
void
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *
gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->type == AUDIO_CODEC_TYPE_NONE);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
}
/**
* gst_basertpaudiopayload_set_frame_options:
* gst_base_rtp_audio_payload_set_frame_options:
* @basertpaudiopayload: a pointer to the element.
* @frame_duration: The duraction of an audio frame in milliseconds.
* @frame_size: The size of an audio frame in bytes.
@ -151,7 +171,7 @@ gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *
*
*/
void
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint frame_duration, gint frame_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
@ -161,7 +181,7 @@ gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
}
/**
* gst_basertpaudiopayload_set_sample_options:
* gst_base_rtp_audio_payload_set_sample_options:
* @basertpaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bytes.
*
@ -169,7 +189,7 @@ gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
*
*/
void
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
@ -178,7 +198,7 @@ gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
}
static GstFlowReturn
gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload,
gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
@ -188,11 +208,11 @@ gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload,
ret = GST_FLOW_ERROR;
if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload,
if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,
buffer);
} else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload,
} else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
ret = gst_base_rtp_audio_payload_handle_sample_based_buffer (basepayload,
buffer);
} else {
GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
@ -203,7 +223,7 @@ gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload,
/* this assumes all frames have a constant duration and a constant size */
static GstFlowReturn
gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
@ -233,8 +253,9 @@ gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
if (!gst_basertppayload_is_filled (basepayload,
gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0),
GST_BUFFER_DURATION (buffer))) {
ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer),
GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer));
ret = gst_base_rtp_audio_payload_push (basepayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
return ret;
@ -273,7 +294,7 @@ gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
/* currently available */
(available / frame_size) * frame_size);
ret = gst_basertpaudiopayload_push (basepayload, data, payload_len,
ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
basertpaudiopayload->base_ts);
gfloat ts_inc = (payload_len * frame_duration) / frame_size;
@ -298,7 +319,7 @@ gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
}
static GstFlowReturn
gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
@ -327,8 +348,9 @@ gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
if (!gst_basertppayload_is_filled (basepayload,
gst_rtp_buffer_calc_packet_len (GST_BUFFER_SIZE (buffer), 0, 0),
GST_BUFFER_DURATION (buffer))) {
ret = gst_basertpaudiopayload_push (basepayload, GST_BUFFER_DATA (buffer),
GST_BUFFER_SIZE (buffer), GST_BUFFER_TIMESTAMP (buffer));
ret = gst_base_rtp_audio_payload_push (basepayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
return ret;
@ -364,7 +386,7 @@ gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
/* currently available */
available);
ret = gst_basertpaudiopayload_push (basepayload, data, payload_len,
ret = gst_base_rtp_audio_payload_push (basepayload, data, payload_len,
basertpaudiopayload->base_ts);
gfloat num = payload_len;
@ -386,7 +408,7 @@ gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
}
static GstFlowReturn
gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
gst_base_rtp_audio_payload_push (GstBaseRTPPayload * basepayload, guint8 * data,
guint payload_len, GstClockTime timestamp)
{
GstBuffer *outbuf;

24
gst-libs/gst/rtp/gstbasertpaudiopayload.h
View file

@ -28,8 +28,10 @@ G_BEGIN_DECLS
typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate;
#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
(gst_basertpaudiopayload_get_type())
(gst_base_rtp_audio_payload_get_type())
#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), \
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
@ -41,14 +43,10 @@ typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
typedef enum {
AUDIO_CODEC_TYPE_NONE,
AUDIO_CODEC_TYPE_FRAME_BASED,
AUDIO_CODEC_TYPE_SAMPLE_BASED
} AudioCodecType;
struct _GstBaseRTPAudioPayload
{
GstBaseRTPAudioPayloadPrivate *priv;
GstBaseRTPPayload payload;
GstClockTime base_ts;
@ -57,8 +55,6 @@ struct _GstBaseRTPAudioPayload
gint sample_size;
AudioCodecType type;
gpointer _gst_reserved[GST_PADDING];
};
@ -69,22 +65,22 @@ struct _GstBaseRTPAudioPayloadClass
gpointer _gst_reserved[GST_PADDING];
};
GType gst_basertpaudiopayload_get_type (void);
GType gst_base_rtp_audio_payload_get_type (void);
void
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload
*basertpaudiopayload);
void
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload
*basertpaudiopayload);
void
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint frame_duration, gint frame_size);
void
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
G_END_DECLS