gstreamer/gst-libs/gst/rtp/gstbasertpaudiopayload.c

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/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
*
* 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.
*/
/**
* SECTION:gstbasertpaudiopayload
* @short_description: Base class for audio RTP payloader
*
* <refsect2>
* <para>
* Provides a base class for audio RTP payloaders for frame or sample based
* audio codecs (constant bitrate)
* </para>
* <para>
* This class derives from GstBaseRTPPayload. It can be used for payloading
* audio codecs. It will only work with constant bitrate codecs. It supports
* both frame based and sample based codecs. It takes care of packing up the
* audio data into RTP packets and filling up the headers accordingly. The
* payloading is done based on the maximum MTU (mtu) and the maximum time per
* packet (max-ptime). The general idea is to divide large data buffers into
* smaller RTP packets. The RTP packet size is the minimum of either the MTU,
* max-ptime (if set) or available data. The RTP packet size is always larger or
* equal to min-ptime (if set). If min-ptime is not set, any residual data is
* sent in a last RTP packet. In the case of frame based codecs, the resulting
* RTP packets always contain full frames.
* </para>
* <title>Usage</title>
* <para>
* To use this base class, your child element needs to call either
* gst_base_rtp_audio_payload_set_frame_based() or
* gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the
* element's _init() function. Then, the child element must call either
* gst_base_rtp_audio_payload_set_frame_options() or
* gst_base_rtp_audio_payload_set_sample_options(). Since GstBaseRTPAudioPayload
* derives from GstBaseRTPPayload, the child element must set any variables or
* call/override any functions required by that base class. The child element
* does not need to override any other functions specific to
* GstBaseRTPAudioPayload.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/base/gstadapter.h>
#include "gstbasertpaudiopayload.h"
GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
typedef enum
{
AUDIO_CODEC_TYPE_NONE,
AUDIO_CODEC_TYPE_FRAME_BASED,
AUDIO_CODEC_TYPE_SAMPLE_BASED
} AudioCodecType;
struct _GstBaseRTPAudioPayloadPrivate
{
AudioCodecType type;
GstAdapter *adapter;
guint64 min_ptime;
};
#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \
(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \
GstBaseRTPAudioPayloadPrivate))
static void gst_base_rtp_audio_payload_finalize (GObject * object);
static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload
* payload, GstBuffer * buffer);
static GstFlowReturn
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
static GstFlowReturn
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer);
static GstStateChangeReturn
gst_base_rtp_payload_audio_change_state (GstElement * element,
GstStateChange transition);
static gboolean
gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event);
GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,
GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_base_rtp_audio_payload_base_init (gpointer klass)
{
}
static void
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_base_rtp_audio_payload_finalize);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state);
gstbasertppayload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer);
gstbasertppayload_class->handle_event =
GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event);
GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,
"base audio RTP payloader");
}
static void
gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,
GstBaseRTPAudioPayloadClass * klass)
{
basertpaudiopayload->priv =
GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);
basertpaudiopayload->base_ts = 0;
basertpaudiopayload->priv->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;
basertpaudiopayload->priv->adapter = gst_adapter_new ();
}
static void
gst_base_rtp_audio_payload_finalize (GObject * object)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
g_object_unref (basertpaudiopayload->priv->adapter);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
/**
* 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
* audio codec
*
*/
void
gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_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
* audio codec
*
*/
void
gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *
basertpaudiopayload)
{
g_return_if_fail (basertpaudiopayload != NULL);
g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);
basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;
}
/**
* 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.
*
* Sets the options for frame based audio codecs.
*
*/
void
gst_base_rtp_audio_payload_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;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
}
/**
* gst_base_rtp_audio_payload_set_sample_options:
* @basertpaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bytes.
*
* Sets the options for sample based audio codecs.
*
*/
void
gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload
* basertpaudiopayload, gint sample_size)
{
g_return_if_fail (basertpaudiopayload != NULL);
basertpaudiopayload->sample_size = sample_size;
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
}
static GstFlowReturn
gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstFlowReturn ret;
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
ret = GST_FLOW_ERROR;
if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {
ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,
buffer);
} 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");
}
return ret;
}
/* this assumes all frames have a constant duration and a constant size */
static GstFlowReturn
gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
const guint8 *data = NULL;
GstFlowReturn ret;
guint available;
gint frame_size, frame_duration;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gboolean use_adapter = FALSE;
guint minptime_ms;
ret = GST_FLOW_OK;
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;
/* 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;
}
}
max_payload_len = 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);
/* min number of bytes based on a given ptime, has to be a multiple
of frame duration */
minptime_ms = basepayload->min_ptime / 1000000;
minptime_octets = frame_size * (int) (minptime_ms / frame_duration);
min_payload_len = MAX (minptime_octets, frame_size);
if (min_payload_len > max_payload_len) {
min_payload_len = max_payload_len;
}
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
if (basertpaudiopayload->priv->adapter &&
gst_adapter_available (basertpaudiopayload->priv->adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
use_adapter = TRUE;
} else {
/* let's set the base timestamp */
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
/* If buffer fits on an RTP packet, let's just push it through */
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
GST_BUFFER_SIZE (buffer) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
return ret;
}
available = GST_BUFFER_SIZE (buffer);
data = (guint8 *) GST_BUFFER_DATA (buffer);
}
/* as long as we have full frames */
while (available >= min_payload_len) {
gfloat ts_inc;
/* We send as much as we can */
payload_len = MIN (max_payload_len, (available / frame_size) * frame_size);
if (use_adapter) {
data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
}
ret =
gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
basertpaudiopayload->base_ts);
ts_inc = (payload_len * frame_duration) / frame_size;
ts_inc = ts_inc * GST_MSECOND;
basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc);
if (use_adapter) {
gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
} else {
available -= payload_len;
data += payload_len;
}
}
if (!use_adapter) {
if (available != 0 && basertpaudiopayload->priv->adapter) {
GstBuffer *buf;
buf = gst_buffer_create_sub (buffer,
GST_BUFFER_SIZE (buffer) - available, available);
gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
} else {
gst_buffer_unref (buffer);
}
}
return ret;
}
static GstFlowReturn
gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *
basepayload, GstBuffer * buffer)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
guint payload_len;
const guint8 *data = NULL;
GstFlowReturn ret;
guint available;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = 0;
guint min_payload_len;
guint max_payload_len;
gboolean use_adapter = FALSE;
guint sample_size;
ret = GST_FLOW_OK;
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;
/* 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);
}
max_payload_len = MIN (
/* MTU max */
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
(basertpaudiopayload), 0, 0),
/* ptime max */
maxptime_octets);
/* min number of bytes based on a given ptime, has to be a multiple
of sample rate */
minptime_octets = basepayload->min_ptime * basepayload->clock_rate /
(sample_size * GST_SECOND);
min_payload_len = MAX (minptime_octets, sample_size);
if (min_payload_len > max_payload_len) {
min_payload_len = max_payload_len;
}
GST_DEBUG_OBJECT (basertpaudiopayload,
"Calculated min_payload_len %u and max_payload_len %u",
min_payload_len, max_payload_len);
if (basertpaudiopayload->priv->adapter &&
gst_adapter_available (basertpaudiopayload->priv->adapter)) {
/* If there is always data in the adapter, we have to use it */
gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
use_adapter = TRUE;
} else {
/* let's set the base timestamp */
basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);
/* If buffer fits on an RTP packet, let's just push it through */
/* this will check against max_ptime and max_mtu */
if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&
GST_BUFFER_SIZE (buffer) <= max_payload_len) {
ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),
GST_BUFFER_TIMESTAMP (buffer));
gst_buffer_unref (buffer);
return ret;
}
available = GST_BUFFER_SIZE (buffer);
data = (guint8 *) GST_BUFFER_DATA (buffer);
}
while (available >= min_payload_len) {
gfloat num, datarate;
payload_len =
MIN (max_payload_len, (available / sample_size) * sample_size);
if (use_adapter) {
data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);
}
ret =
gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,
basertpaudiopayload->base_ts);
num = payload_len;
datarate = (sample_size * basepayload->clock_rate);
basertpaudiopayload->base_ts +=
/* payload_len (bytes) * nsecs/sec / datarate (bytes*sec) */
gst_gdouble_to_guint64 (num / datarate * GST_SECOND);
GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT,
GST_TIME_ARGS (basertpaudiopayload->base_ts));
if (use_adapter) {
gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);
available = gst_adapter_available (basertpaudiopayload->priv->adapter);
} else {
available -= payload_len;
data += payload_len;
}
}
if (!use_adapter) {
if (available != 0 && basertpaudiopayload->priv->adapter) {
GstBuffer *buf;
buf = gst_buffer_create_sub (buffer,
GST_BUFFER_SIZE (buffer) - available, available);
gst_adapter_push (basertpaudiopayload->priv->adapter, buf);
} else {
gst_buffer_unref (buffer);
}
}
return ret;
}
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
/**
* gst_base_rtp_audio_payload_push:
* @baseaudiopayload: a #GstBaseRTPPayload
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
* @data: data to set as payload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of @data as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* Returns: a #GstFlowReturn
*
* Since: 0.10.13
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
*/
GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstBaseRTPPayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
basepayload = GST_BASE_RTP_PAYLOAD (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);
/* 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;
}
static GstStateChangeReturn
gst_base_rtp_payload_audio_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPAudioPayload *basertppayload;
GstStateChangeReturn ret;
basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element);
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (basertppayload->priv->adapter) {
gst_adapter_clear (basertppayload->priv->adapter);
}
break;
default:
break;
}
return ret;
}
static gboolean
gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
gboolean res = FALSE;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
break;
case GST_EVENT_FLUSH_STOP:
if (basertpaudiopayload->priv->adapter) {
gst_adapter_clear (basertpaudiopayload->priv->adapter);
}
break;
default:
break;
}
gst_object_unref (basertpaudiopayload);
/* return FALSE to let parent handle the remainder of the event */
return res;
}
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
/**
* gst_base_rtp_audio_payload_get_adapter:
* @basertpaudiopayload: a #GstBaseRTPAudioPayload
*
* Gets the internal adapter used by the depayloader.
*
* Returns: a #GstAdapter.
*
* Since: 0.10.13
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
*/
GstAdapter *
gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload
* basertpaudiopayload)
{
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
GstAdapter *adapter;
if ((adapter = basertpaudiopayload->priv->adapter))
g_object_ref (adapter);
Add RTCP docs. Original commit message from CVS: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: (gst_base_rtp_audio_payload_get_adapter): Add RTCP docs. Fix some more docs. * gst-libs/gst/rtp/Makefile.am: * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data), (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate), (gst_rtcp_buffer_get_packet_count), (read_packet_header), (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next), (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove), (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type), (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length), (gst_rtcp_packet_sr_get_sender_info), (gst_rtcp_packet_sr_set_sender_info), (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb), (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb), (gst_rtcp_packet_sdes_get_chunk_count), (gst_rtcp_packet_sdes_first_chunk), (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc), (gst_rtcp_packet_sdes_first_item), (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item), (gst_rtcp_packet_bye_get_ssrc_count), (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc), (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset), (gst_rtcp_packet_bye_get_reason_len), (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason): * gst-libs/gst/rtp/gstrtcpbuffer.h: Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
return adapter;
}