gstreamer/gst/inter/gstinteraudiosink.c

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/* 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-gstinteraudiosink
*
* The interaudiosink element is an audio sink element. It is used
* in connection with a interaudiosrc element in a different pipeline,
* similar to intervideosink and intervideosrc.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v audiotestsrc ! queue ! interaudiosink
* ]|
*
* The interaudiosink element cannot be used effectively with gst-launch,
* as it requires a second pipeline in the application to receive the
* 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/gstbasesink.h>
#include <gst/audio/audio.h>
#include "gstinteraudiosink.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
/* prototypes */
static void gst_inter_audio_sink_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_inter_audio_sink_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_inter_audio_sink_finalize (GObject * object);
static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
GstCaps * caps);
static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
GstEvent * event);
static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
static gboolean gst_inter_audio_sink_query (GstBaseSink * sink,
GstQuery * query);
enum
{
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PROP_0,
PROP_CHANNEL
};
#define DEFAULT_CHANNEL ("default")
/* pad templates */
static GstStaticPadTemplate gst_inter_audio_sink_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL))
);
/* class initialization */
#define parent_class gst_inter_audio_sink_parent_class
G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
static void
gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category,
"interaudiosink", 0, "debug category for interaudiosink element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_inter_audio_sink_sink_template));
gst_element_class_set_static_metadata (element_class,
"Internal audio sink",
"Sink/Audio",
"Virtual audio sink for internal process communication",
"David Schleef <ds@schleef.org>");
gobject_class->set_property = gst_inter_audio_sink_set_property;
gobject_class->get_property = gst_inter_audio_sink_get_property;
gobject_class->finalize = gst_inter_audio_sink_finalize;
base_sink_class->get_times =
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
base_sink_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_query);
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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_CHANNEL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink)
{
interaudiosink->channel = g_strdup (DEFAULT_CHANNEL);
interaudiosink->input_adapter = gst_adapter_new ();
}
void
gst_inter_audio_sink_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
switch (property_id) {
case PROP_CHANNEL:
g_free (interaudiosink->channel);
interaudiosink->channel = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_sink_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
switch (property_id) {
case PROP_CHANNEL:
g_value_set_string (value, interaudiosink->channel);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_inter_audio_sink_finalize (GObject * object)
{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
/* clean up object here */
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g_free (interaudiosink->channel);
gst_object_unref (interaudiosink->input_adapter);
G_OBJECT_CLASS (gst_inter_audio_sink_parent_class)->finalize (object);
}
static void
gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
*start = GST_BUFFER_TIMESTAMP (buffer);
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
*end = *start + GST_BUFFER_DURATION (buffer);
} else {
if (interaudiosink->info.rate > 0) {
*end = *start +
gst_util_uint64_scale_int (gst_buffer_get_size (buffer), GST_SECOND,
interaudiosink->info.rate * interaudiosink->info.bpf);
}
}
}
}
static gboolean
gst_inter_audio_sink_start (GstBaseSink * sink)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG_OBJECT (interaudiosink, "start");
interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
g_mutex_lock (&interaudiosink->surface->mutex);
memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
/* We want to write latency-time before syncing has happened */
/* FIXME: The other side can change this value when it starts */
gst_base_sink_set_render_delay (sink,
interaudiosink->surface->audio_latency_time);
g_mutex_unlock (&interaudiosink->surface->mutex);
return TRUE;
}
static gboolean
gst_inter_audio_sink_stop (GstBaseSink * sink)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GST_DEBUG_OBJECT (interaudiosink, "stop");
g_mutex_lock (&interaudiosink->surface->mutex);
gst_adapter_clear (interaudiosink->surface->audio_adapter);
memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
g_mutex_unlock (&interaudiosink->surface->mutex);
gst_inter_surface_unref (interaudiosink->surface);
interaudiosink->surface = NULL;
gst_adapter_clear (interaudiosink->input_adapter);
return TRUE;
}
static gboolean
gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_ERROR_OBJECT (sink, "Failed to parse caps %" GST_PTR_FORMAT, caps);
return FALSE;
}
g_mutex_lock (&interaudiosink->surface->mutex);
interaudiosink->surface->audio_info = info;
interaudiosink->info = info;
/* TODO: Ideally we would drain the source here */
gst_adapter_clear (interaudiosink->surface->audio_adapter);
g_mutex_unlock (&interaudiosink->surface->mutex);
return TRUE;
}
static gboolean
gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:{
GstBuffer *tmp;
guint n;
if ((n = gst_adapter_available (interaudiosink->input_adapter)) > 0) {
g_mutex_lock (&interaudiosink->surface->mutex);
tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
g_mutex_unlock (&interaudiosink->surface->mutex);
}
break;
}
default:
break;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
static GstFlowReturn
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
guint n, bpf;
guint64 period_time, buffer_time;
guint64 period_samples, buffer_samples;
GST_DEBUG_OBJECT (interaudiosink, "render %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
bpf = interaudiosink->info.bpf;
g_mutex_lock (&interaudiosink->surface->mutex);
buffer_time = interaudiosink->surface->audio_buffer_time;
period_time = interaudiosink->surface->audio_period_time;
if (buffer_time < period_time) {
GST_ERROR_OBJECT (interaudiosink,
"Buffer time smaller than period time (%" GST_TIME_FORMAT " < %"
GST_TIME_FORMAT ")", GST_TIME_ARGS (buffer_time),
GST_TIME_ARGS (period_time));
g_mutex_unlock (&interaudiosink->surface->mutex);
return GST_FLOW_ERROR;
}
buffer_samples =
gst_util_uint64_scale (buffer_time, interaudiosink->info.rate,
GST_SECOND);
period_samples =
gst_util_uint64_scale (period_time, interaudiosink->info.rate,
GST_SECOND);
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / bpf;
while (n > buffer_samples) {
GST_DEBUG_OBJECT (interaudiosink, "flushing %" GST_TIME_FORMAT,
GST_TIME_ARGS (period_time));
gst_adapter_flush (interaudiosink->surface->audio_adapter,
period_samples * bpf);
n -= period_samples;
}
n = gst_adapter_available (interaudiosink->input_adapter);
if (period_samples * bpf > gst_buffer_get_size (buffer) + n) {
gst_adapter_push (interaudiosink->input_adapter, gst_buffer_ref (buffer));
} else {
GstBuffer *tmp;
if (n > 0) {
tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
}
g_mutex_unlock (&interaudiosink->surface->mutex);
return GST_FLOW_OK;
}
static gboolean
gst_inter_audio_sink_query (GstBaseSink * sink, GstQuery * query)
{
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
gboolean ret;
GST_DEBUG_OBJECT (sink, "query");
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (sink, "latency query");
if ((ret =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (sink), &live,
&us_live, &min_l, &max_l))) {
GstClockTime base_latency, min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
/* FIXME: The other side can change this value when it starts */
base_latency = interaudiosink->surface->audio_latency_time;
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = base_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
GST_DEBUG_OBJECT (sink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (sink,
"peer max %" GST_TIME_FORMAT ", our max latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
GST_TIME_ARGS (max_latency));
} else {
GST_DEBUG_OBJECT (sink,
"peer or we are not live, don't care about latency");
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
ret =
GST_BASE_SINK_CLASS (gst_inter_audio_sink_parent_class)->query (sink,
query);
break;
}
return ret;
}