gstreamer/gst/inter/gstinteraudiosrc.c
Sebastian Dröge 860ccd414d Merge remote-tracking branch 'origin/0.10'
Conflicts:
	NEWS
	RELEASE
	common
	configure.ac
	docs/libs/gst-plugins-bad-libs-sections.txt
	docs/plugins/gst-plugins-bad-plugins.args
	docs/plugins/gst-plugins-bad-plugins.hierarchy
	docs/plugins/gst-plugins-bad-plugins.interfaces
	docs/plugins/inspect/plugin-adpcmdec.xml
	docs/plugins/inspect/plugin-adpcmenc.xml
	docs/plugins/inspect/plugin-assrender.xml
	docs/plugins/inspect/plugin-audiovisualizers.xml
	docs/plugins/inspect/plugin-autoconvert.xml
	docs/plugins/inspect/plugin-bayer.xml
	docs/plugins/inspect/plugin-bz2.xml
	docs/plugins/inspect/plugin-camerabin2.xml
	docs/plugins/inspect/plugin-celt.xml
	docs/plugins/inspect/plugin-dataurisrc.xml
	docs/plugins/inspect/plugin-debugutilsbad.xml
	docs/plugins/inspect/plugin-dtmf.xml
	docs/plugins/inspect/plugin-dtsdec.xml
	docs/plugins/inspect/plugin-dvbsuboverlay.xml
	docs/plugins/inspect/plugin-dvdspu.xml
	docs/plugins/inspect/plugin-faac.xml
	docs/plugins/inspect/plugin-faad.xml
	docs/plugins/inspect/plugin-gsm.xml
	docs/plugins/inspect/plugin-h264parse.xml
	docs/plugins/inspect/plugin-mms.xml
	docs/plugins/inspect/plugin-modplug.xml
	docs/plugins/inspect/plugin-mpeg2enc.xml
	docs/plugins/inspect/plugin-mpegdemux2.xml
	docs/plugins/inspect/plugin-mpegtsdemux.xml
	docs/plugins/inspect/plugin-mpegvideoparse.xml
	docs/plugins/inspect/plugin-mplex.xml
	docs/plugins/inspect/plugin-pcapparse.xml
	docs/plugins/inspect/plugin-rawparse.xml
	docs/plugins/inspect/plugin-rtpmux.xml
	docs/plugins/inspect/plugin-rtpvp8.xml
	docs/plugins/inspect/plugin-scaletempo.xml
	docs/plugins/inspect/plugin-schro.xml
	docs/plugins/inspect/plugin-sdp.xml
	docs/plugins/inspect/plugin-segmentclip.xml
	docs/plugins/inspect/plugin-shm.xml
	docs/plugins/inspect/plugin-videomaxrate.xml
	docs/plugins/inspect/plugin-videoparsersbad.xml
	docs/plugins/inspect/plugin-vp8.xml
	docs/plugins/inspect/plugin-y4mdec.xml
	ext/celt/gstceltdec.c
	ext/dts/gstdtsdec.c
	ext/modplug/gstmodplug.cc
	ext/opus/gstopusenc.c
	gst-libs/gst/video/gstbasevideocodec.c
	gst-libs/gst/video/gstbasevideocodec.h
	gst-libs/gst/video/gstbasevideodecoder.c
	gst-libs/gst/video/gstbasevideodecoder.h
	gst-libs/gst/video/gstbasevideoencoder.c
	gst-libs/gst/video/gstbasevideoencoder.h
	gst/adpcmdec/Makefile.am
	gst/audiovisualizers/gstbaseaudiovisualizer.c
	gst/h264parse/gsth264parse.c
	gst/mpegdemux/mpegtsparse.c
	gst/mpegtsdemux/mpegtsbase.c
	gst/mpegtsdemux/mpegtspacketizer.c
	gst/mpegtsdemux/mpegtsparse.c
	gst/mpegtsdemux/tsdemux.c
	gst/mpegtsdemux/tsdemux.h
	gst/mxf/mxfdemux.c
	gst/rawparse/gstaudioparse.c
	gst/videoparsers/gsth263parse.c
	gst/videoparsers/gsth264parse.c
	sys/d3dvideosink/d3dvideosink.c
	sys/decklink/gstdecklinksink.cpp
	sys/dvb/gstdvbsrc.c
	sys/shm/gstshmsrc.c
	sys/vdpau/h264/gstvdph264dec.c
	sys/vdpau/mpeg/gstvdpmpegdec.c
	tests/examples/opencv/gst_element_print_properties.c
	win32/common/config.h
2012-03-29 17:41:53 +02:00

534 lines
15 KiB
C

/* GStreamer
* Copyright (C) 2011 David A. Schleef <ds@schleef.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., 51 Franklin Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-gstinteraudiosrc
*
* The interaudiosrc element is an audio source element. It is used
* in connection with a interaudiosink element in a different pipeline.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v interaudiosrc ! queue ! audiosink
* ]|
*
* The interaudiosrc element cannot be used effectively with gst-launch,
* as it requires a second pipeline in the application to send audio.
* See the gstintertest.c example in the gst-plugins-bad source code for
* more details.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasesrc.h>
#include "gstinteraudiosrc.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
/* prototypes */
static void gst_inter_audio_src_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_inter_audio_src_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_inter_audio_src_dispose (GObject * object);
static void gst_inter_audio_src_finalize (GObject * object);
static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src);
static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
static gboolean gst_inter_audio_src_negotiate (GstBaseSrc * src);
static gboolean gst_inter_audio_src_newsegment (GstBaseSrc * src);
static gboolean gst_inter_audio_src_start (GstBaseSrc * src);
static gboolean gst_inter_audio_src_stop (GstBaseSrc * src);
static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end);
static gboolean gst_inter_audio_src_is_seekable (GstBaseSrc * src);
static gboolean gst_inter_audio_src_unlock (GstBaseSrc * src);
static gboolean gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event);
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf);
static gboolean gst_inter_audio_src_do_seek (GstBaseSrc * src,
GstSegment * segment);
static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query);
static gboolean gst_inter_audio_src_check_get_range (GstBaseSrc * src);
static void gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
static gboolean gst_inter_audio_src_unlock_stop (GstBaseSrc * src);
static gboolean
gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek,
GstSegment * segment);
enum
{
PROP_0,
PROP_CHANNEL
};
/* pad templates */
static GstStaticPadTemplate gst_inter_audio_src_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 16, "
"depth = (int) 16, " "rate = (int) 48000, " "channels = (int) 2")
);
/* class initialization */
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc", 0, \
"debug category for interaudiosrc element");
GST_BOILERPLATE_FULL (GstInterAudioSrc, gst_inter_audio_src, GstBaseSrc,
GST_TYPE_BASE_SRC, DEBUG_INIT);
static void
gst_inter_audio_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_inter_audio_src_src_template));
gst_element_class_set_details_simple (element_class,
"Internal audio source",
"Source/Audio",
"Virtual audio source for internal process communication",
"David Schleef <ds@schleef.org>");
}
static void
gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
gobject_class->set_property = gst_inter_audio_src_set_property;
gobject_class->get_property = gst_inter_audio_src_get_property;
gobject_class->dispose = gst_inter_audio_src_dispose;
gobject_class->finalize = gst_inter_audio_src_finalize;
base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps);
base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
if (0)
base_src_class->negotiate =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_negotiate);
base_src_class->newsegment =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_newsegment);
base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start);
base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop);
base_src_class->get_times = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_times);
if (0)
base_src_class->is_seekable =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_is_seekable);
base_src_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock);
base_src_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_src_event);
base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create);
if (0)
base_src_class->do_seek = GST_DEBUG_FUNCPTR (gst_inter_audio_src_do_seek);
base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query);
if (0)
base_src_class->check_get_range =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_check_get_range);
base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate);
if (0)
base_src_class->unlock_stop =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_unlock_stop);
if (0)
base_src_class->prepare_seek_segment =
GST_DEBUG_FUNCPTR (gst_inter_audio_src_prepare_seek_segment);
#if 0
g_object_class_install_property (gobject_class, PROP_CHANNEL,
g_param_spec_string ("channel", "Channel",
"Channel name to match inter src and sink elements",
"default", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
}
static void
gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc,
GstInterAudioSrcClass * interaudiosrc_class)
{
gst_base_src_set_live (GST_BASE_SRC (interaudiosrc), TRUE);
gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1);
interaudiosrc->channel = g_strdup ("default");
}
void
gst_inter_audio_src_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
switch (property_id) {
case PROP_CHANNEL:
g_free (interaudiosrc->channel);
interaudiosrc->channel = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_src_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
switch (property_id) {
case PROP_CHANNEL:
g_value_set_string (value, interaudiosrc->channel);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_src_dispose (GObject * object)
{
/* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
/* clean up as possible. may be called multiple times */
G_OBJECT_CLASS (parent_class)->dispose (object);
}
void
gst_inter_audio_src_finalize (GObject * object)
{
/* GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object); */
/* clean up object here */
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_inter_audio_src_get_caps (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
return NULL;
}
static gboolean
gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
const GstStructure *structure;
gboolean ret;
int sample_rate;
GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &sample_rate);
if (ret) {
interaudiosrc->sample_rate = sample_rate;
}
return ret;
}
static gboolean
gst_inter_audio_src_negotiate (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "negotiate");
return TRUE;
}
static gboolean
gst_inter_audio_src_newsegment (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "newsegment");
return TRUE;
}
static gboolean
gst_inter_audio_src_start (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "start");
interaudiosrc->surface = gst_inter_surface_get (interaudiosrc->channel);
return TRUE;
}
static gboolean
gst_inter_audio_src_stop (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "stop");
gst_inter_surface_unref (interaudiosrc->surface);
interaudiosrc->surface = NULL;
return TRUE;
}
static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "get_times");
/* for live sources, sync on the timestamp of the buffer */
if (gst_base_src_is_live (src)) {
GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* get duration to calculate end time */
GstClockTime duration = GST_BUFFER_DURATION (buffer);
if (GST_CLOCK_TIME_IS_VALID (duration)) {
*end = timestamp + duration;
}
*start = timestamp;
}
} else {
*start = -1;
*end = -1;
}
}
static gboolean
gst_inter_audio_src_is_seekable (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "is_seekable");
return FALSE;
}
static gboolean
gst_inter_audio_src_unlock (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "unlock");
return TRUE;
}
static gboolean
gst_inter_audio_src_event (GstBaseSrc * src, GstEvent * event)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
gboolean ret;
GST_DEBUG_OBJECT (interaudiosrc, "event");
switch (GST_EVENT_TYPE (event)) {
default:
ret = GST_BASE_SRC_CLASS (parent_class)->event (src, event);
}
return ret;
}
#define SIZE 1600
static GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstBuffer *buffer;
int n;
GST_DEBUG_OBJECT (interaudiosrc, "create");
buffer = NULL;
g_mutex_lock (interaudiosrc->surface->mutex);
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / 4;
if (n > SIZE * 2) {
GST_DEBUG ("flushing %d samples", SIZE / 2);
gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * 4);
n -= (SIZE / 2);
}
if (n > SIZE)
n = SIZE;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * 4);
}
g_mutex_unlock (interaudiosrc->surface->mutex);
if (n < SIZE) {
GstBuffer *newbuf = gst_buffer_new_and_alloc (SIZE * 4);
GST_DEBUG ("creating %d samples of silence", SIZE - n);
memset (GST_BUFFER_DATA (newbuf) + n * 4, 0, SIZE * 4 - n * 4);
if (buffer) {
memcpy (GST_BUFFER_DATA (newbuf), GST_BUFFER_DATA (buffer), n * 4);
gst_buffer_unref (buffer);
}
buffer = newbuf;
}
n = SIZE;
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->sample_rate);
GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) =
gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
interaudiosrc->sample_rate) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = -1;
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (interaudiosrc->n_samples == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
}
gst_buffer_set_caps (buffer, GST_PAD_CAPS (GST_BASE_SRC_PAD (interaudiosrc)));
interaudiosrc->n_samples += n;
*buf = buffer;
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_src_do_seek (GstBaseSrc * src, GstSegment * segment)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "do_seek");
return FALSE;
}
static gboolean
gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
gboolean ret;
GST_DEBUG_OBJECT (interaudiosrc, "query");
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
GstClockTime min_latency, max_latency;
min_latency = 30 * gst_util_uint64_scale_int (GST_SECOND, SIZE, 48000);
max_latency = min_latency;
GST_ERROR_OBJECT (src,
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query,
gst_base_src_is_live (src), min_latency, max_latency);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
break;
}
return ret;
}
static gboolean
gst_inter_audio_src_check_get_range (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "get_range");
return FALSE;
}
static void
gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
GST_DEBUG_OBJECT (interaudiosrc, "fixate");
gst_structure_fixate_field_nearest_int (structure, "channels", 2);
gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
}
static gboolean
gst_inter_audio_src_unlock_stop (GstBaseSrc * src)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "stop");
return TRUE;
}
static gboolean
gst_inter_audio_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * seek,
GstSegment * segment)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GST_DEBUG_OBJECT (interaudiosrc, "seek_segment");
return FALSE;
}