gstreamer/gst/inter/gstinteraudiosrc.c
Tim-Philipp Müller e861c72efc interaudiosrc: make silence memory actually contain silence
instead of random data. Reported by Marco Micheletti on
gstreamer-devel.
2013-08-14 18:19:21 +01:00

384 lines
11 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 "gstinteraudiosrc.h"
#include <gst/gst.h>
#include <gst/base/gstbasesrc.h>
#include <gst/audio/audio.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_finalize (GObject * object);
static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
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 GstFlowReturn
gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf);
static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query);
static GstCaps *gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
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, format = (string) " GST_AUDIO_NE (S16) ", "
"rate = (int) 48000, channels = (int) 2")
);
/* class initialization */
G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_SRC);
static void
gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc",
0, "debug category for interaudiosrc element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_inter_audio_src_src_template));
gst_element_class_set_static_metadata (element_class,
"Internal audio source",
"Source/Audio",
"Virtual audio source for internal process communication",
"David Schleef <ds@schleef.org>");
gobject_class->set_property = gst_inter_audio_src_set_property;
gobject_class->get_property = gst_inter_audio_src_get_property;
gobject_class->finalize = gst_inter_audio_src_finalize;
base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
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);
base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create);
base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query);
base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate);
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));
}
static void
gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc)
{
gst_base_src_set_format (GST_BASE_SRC (interaudiosrc), GST_FORMAT_TIME);
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_finalize (GObject * object)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
/* clean up object here */
g_free (interaudiosrc->channel);
G_OBJECT_CLASS (gst_inter_audio_src_parent_class)->finalize (object);
}
static gboolean
gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
const GstStructure *structure;
GstAudioInfo info;
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;
ret = gst_pad_set_caps (src->srcpad, caps);
}
if (gst_audio_info_from_caps (&info, caps)) {
interaudiosrc->finfo = info.finfo;
}
return ret;
}
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;
interaudiosrc->finfo = NULL;
return TRUE;
}
static void
gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GST_DEBUG_OBJECT (src, "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;
}
}
#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 * 3) {
GST_WARNING ("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);
} else {
buffer = gst_buffer_new ();
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (n < SIZE) {
GstMapInfo map;
GstMemory *mem;
GST_WARNING ("creating %d samples of silence", SIZE - n);
mem = gst_allocator_alloc (NULL, (SIZE - n) * 4, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_fill_silence (interaudiosrc->finfo, map.data, map.size);
gst_memory_unmap (mem, &map);
}
buffer = gst_buffer_make_writable (buffer);
gst_buffer_prepend_memory (buffer, mem);
}
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);
}
interaudiosrc->n_samples += n;
*buf = buffer;
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query)
{
gboolean ret;
GST_DEBUG_OBJECT (src, "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 (gst_inter_audio_src_parent_class)->query (src,
query);
break;
}
return ret;
}
static GstCaps *
gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
{
GstStructure *structure;
GST_DEBUG_OBJECT (src, "fixate");
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (structure, "channels", 2);
gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
return caps;
}