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gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children

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Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
This commit is contained in:
Philippe Kalaf 2006-04-13 03:55:12 +00:00
parent d18d5d33d0
commit a916af7c48
5 changed files with 529 additions and 0 deletions
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8
ChangeLog
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@ -1,3 +1,11 @@
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
2006-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac:

2
gst-libs/gst/rtp/Makefile.am
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@ -2,12 +2,14 @@ libgstrtpincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtp
libgstrtpinclude_HEADERS = gstrtpbuffer.h \
gstbasertppayload.h \
gstbasertpaudiopayload.h \
gstbasertpdepayload.h
lib_LTLIBRARIES = libgstrtp-@GST_MAJORMINOR@.la
libgstrtp_@GST_MAJORMINOR@_la_SOURCES = gstrtpbuffer.c \
gstbasertppayload.c \
gstbasertpaudiopayload.c \
gstbasertpdepayload.c
libgstrtp_@GST_MAJORMINOR@_la_CFLAGS = $(GST_CFLAGS)

429
gst-libs/gst/rtp/gstbasertpaudiopayload.c Normal file
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@ -0,0 +1,429 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <math.h>
#include "gstbasertpaudiopayload.h"
GST_DEBUG_CATEGORY (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
/* let us define a minimum of 10 ms for sample based codecs */
#define GST_RTP_MIN_PTIME_MS 10
static void gst_basertpaudiopayload_finalize (GObject * object);
static GstFlowReturn
gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
guint payload_len, GstClockTime timestamp);
static GstFlowReturn gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static GstFlowReturn
gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
static GstFlowReturn
gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_basertpaudiopayload,
GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_basertpaudiopayload_base_init (gpointer klass)
{
}
static void
gst_basertpaudiopayload_class_init (GstBaseRTPAudioPayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize = gst_basertpaudiopayload_finalize;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->handle_buffer =
gst_basertpaudiopayload_handle_buffer;
GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
"base audio RTP payloader");
}
static void
gst_basertpaudiopayload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
GstBaseRTPAudioPayloadClass * klass)
{
basertpaudiopayload->adapter = gst_adapter_new ();
basertpaudiopayload->adapter_base_ts = 0;
basertpaudiopayload->type = AUDIO_CODEC_TYPE_NONE;
/* these need to be set by child object if frame based */
basertpaudiopayload->frame_size = 0;
basertpaudiopayload->frame_duration = 0;
/* these need to be set by child object if sample based */
basertpaudiopayload->sample_size = 0;
}
static void
gst_basertpaudiopayload_finalize (GObject * object)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
g_object_unref (basertpaudiopayload->adapter);
basertpaudiopayload->adapter = NULL;
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
void
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) {
GST_ERROR_OBJECT (basertpaudiopayload,
"Codec type already set! You should only set this once!");
}
basertpaudiopayload->type = AUDIO_CODEC_TYPE_FRAME_BASED;
}
void
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
if (basertpaudiopayload->type != AUDIO_CODEC_TYPE_NONE) {
GST_ERROR_OBJECT (basertpaudiopayload,
"Codec type already set! You should only set this once!");
}
basertpaudiopayload->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
}
/* These are options that need to be set for frame based audio codecs */
void
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint frame_duration, gint frame_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->frame_size = frame_size;
basertpaudiopayload->frame_duration = frame_duration;
}
void
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->sample_size = sample_size;
}
static GstFlowReturn
gst_basertpaudiopayload_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
ret = GST_FLOW_ERROR;
if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
ret = gst_basertpaudiopayload_handle_frame_based_buffer (basepayload,
buffer);
} else if (basertpaudiopayload->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {
ret = gst_basertpaudiopayload_handle_sample_based_buffer (basepayload,
buffer);
} else {
GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");
}
return ret;
}
/* this assumes all frames have a constant duration and a constant size */
static GstFlowReturn
gst_basertpaudiopayload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
guint8 *data;
GstFlowReturn ret;
guint available;
gint frame_size, frame_duration;
guint maxptime_octets = G_MAXUINT;
ret = GST_FLOW_ERROR;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
if (basertpaudiopayload->frame_size == 0 ||
basertpaudiopayload->frame_duration == 0) {
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
frame_size = basertpaudiopayload->frame_size;
frame_duration = basertpaudiopayload->frame_duration;
/* If buffer fits on an RTP packet, let's just push it through without using
* the adapter */
/* this will check again max_ptime and max_mtu */
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));
gst_buffer_unref (buffer);
return ret;
}
/* TODO : would be nice if we had some property that told the payloader to put
* just 1 frame per RTP packet, for the moment we can set the ptime to 0 or
* something smaller or equal to a frame duration */
/* max number of bytes based on given ptime, has to be multiple of
* frame_duration */
if (basepayload->max_ptime != -1) {
guint ptime_ms = basepayload->max_ptime / 1000000;
maxptime_octets = frame_size * (int) (ptime_ms / frame_duration);
if (maxptime_octets == 0) {
GST_WARNING_OBJECT (basertpaudiopayload,
"Given ptime %d is smaller than minimum %d ms, overwriting to minimum",
ptime_ms, frame_duration);
maxptime_octets = frame_size;
}
}
/* if the adapter is empty (should be), let's set the base timestamp */
if (gst_adapter_available (basertpaudiopayload->adapter) == 0) {
basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer);
} else {
GST_ERROR_OBJECT (basertpaudiopayload,
"Adapter should be empty but is not!");
return GST_FLOW_ERROR;
}
gst_adapter_push (basertpaudiopayload->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->adapter);
/* as long as we have full frames */
/* this loop will always empty the adapter till the last frame */
/* TODO Make it possible to set a minimum size per packet, this way the
* algorithm doesn't empty the adapter if there is too little data left and
* will wait until the next buffers to arrive */
while (available >= frame_size) {
/* we need to see how many frames we can get based on maximum MTU, maximum
* ptime and the number of bytes available in the adapter */
payload_len = MIN (MIN (
/* MTU max */
(int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0) / frame_size) * frame_size,
/* ptime max */
maxptime_octets),
/* currently available */
floor (available / frame_size) * frame_size);
data =
(guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
ret =
gst_basertpaudiopayload_push (basepayload, data, payload_len,
basertpaudiopayload->adapter_base_ts);
gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
gfloat ts_inc = (payload_len * frame_duration) / frame_size;
ts_inc = ts_inc * GST_MSECOND;
basertpaudiopayload->adapter_base_ts += ts_inc;
GST_DEBUG_OBJECT (basertpaudiopayload, "%f %f %d", ts_inc,
ts_inc * GST_MSECOND, (payload_len * frame_duration) / frame_size);
GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
available = gst_adapter_available (basertpaudiopayload->adapter);
}
/* adapter should be freed by now */
if (available != 0) {
GST_ERROR_OBJECT (basertpaudiopayload,
"Adapter should be empty but is not!");
return GST_FLOW_ERROR;
}
return ret;
}
static GstFlowReturn
gst_basertpaudiopayload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
guint8 *data;
GstFlowReturn ret;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint sample_size;
ret = GST_FLOW_ERROR;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
if (basertpaudiopayload->sample_size == 0) {
GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
sample_size = basertpaudiopayload->sample_size;
/* If buffer fits on an RTP packet, let's just push it through without using
* the adapter */
/* this will check again max_ptime and max_mtu */
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));
gst_buffer_unref (buffer);
return ret;
}
/* max number of bytes based on given ptime */
if (basepayload->max_ptime != -1) {
maxptime_octets = basepayload->max_ptime * basepayload->clock_rate /
(sample_size * GST_SECOND);
minptime_octets = GST_RTP_MIN_PTIME_MS * basepayload->clock_rate /
(sample_size * 1000);
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated max_octects %u and min_octets %u", maxptime_octets,
minptime_octets);
if (maxptime_octets < minptime_octets) {
GST_WARNING_OBJECT (basertpaudiopayload,
"Given ptime %d is smaller than minimum %d, replacing by %d",
maxptime_octets, minptime_octets, minptime_octets);
maxptime_octets = minptime_octets;
}
}
/* if the adapter is empty (should be), let's set the base timestamp */
if (gst_adapter_available (basertpaudiopayload->adapter) == 0) {
basertpaudiopayload->adapter_base_ts = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (basertpaudiopayload, "Setting to %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
}
gst_adapter_push (basertpaudiopayload->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->adapter);
/* as long as we have full frames */
/* this loop will always empty the adapter till the last frame */
/* TODO Make it possible to set a minimum size per packet, this way the
* algorithm doesn't empty the adapter if there is too little data left and
* will wait until the next buffers to arrive */
while (available >= minptime_octets) {
/* we need to see how many frames we can get based on maximum MTU, maximum
* ptime and the number of bytes available in the adapter */
payload_len = MIN (MIN (
/* MTU max */
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0),
/* ptime max */
maxptime_octets),
/* currently available */
available);
data =
(guint8 *) gst_adapter_peek (basertpaudiopayload->adapter, payload_len);
GST_DEBUG_OBJECT (basertpaudiopayload, "Pushing with ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
ret =
gst_basertpaudiopayload_push (basepayload, data, payload_len,
basertpaudiopayload->adapter_base_ts);
gst_adapter_flush (basertpaudiopayload->adapter, payload_len);
gfloat num = payload_len;
gfloat datarate = (sample_size * basepayload->clock_rate);
basertpaudiopayload->adapter_base_ts +=
/* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */
num / datarate * GST_SECOND;
GST_DEBUG_OBJECT (basertpaudiopayload, "Calculating ts inc %f %f %f", num,
datarate, num / datarate * GST_SECOND);
GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
GST_TIME_ARGS (basertpaudiopayload->adapter_base_ts));
available = gst_adapter_available (basertpaudiopayload->adapter);
}
return ret;
}
static GstFlowReturn
gst_basertpaudiopayload_push (GstBaseRTPPayload * basepayload, guint8 * data,
guint payload_len, GstClockTime timestamp)
{
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt);
payload = gst_rtp_buffer_get_payload (outbuf);
memcpy (payload, data, payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}

88
gst-libs/gst/rtp/gstbasertpaudiopayload.h Normal file
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@ -0,0 +1,88 @@
/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__
#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
(gst_basertpaudiopayload_get_type())
#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))
#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))
#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(obj) \
(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
{
GstBaseRTPPayload payload;
GstClockTime adapter_base_ts;
GstAdapter *adapter;
gint frame_size;
gint frame_duration;
gint sample_size;
AudioCodecType type;
};
struct _GstBaseRTPAudioPayloadClass
{
GstBaseRTPPayloadClass parent_class;
};
gboolean gst_basertpaudiopayload_plugin_init (GstPlugin * plugin);
GType gst_basertpaudiopayload_get_type (void);
void
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
void
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
void
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint frame_duration, gint frame_size);
void
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
G_END_DECLS
#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */

2
gst-libs/gst/rtp/gstrtpbuffer.h
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@ -71,6 +71,8 @@ typedef enum
#define GST_RTP_PAYLOAD_MPV_STRING "32"
#define GST_RTP_PAYLOAD_H263_STRING "34"
#define GST_RTP_PAYLOAD_DYNAMIC_STRING "[96, 127]"
/* creating buffers */
GstBuffer* gst_rtp_buffer_new (void);
void gst_rtp_buffer_allocate_data (GstBuffer *buffer, guint payload_len,